diff options
105 files changed, 3500 insertions, 6106 deletions
diff --git a/CleanSpec.mk b/CleanSpec.mk new file mode 100644 index 0000000..e6d9ebf --- /dev/null +++ b/CleanSpec.mk @@ -0,0 +1,52 @@ +# Copyright (C) 2012 The Android Open Source Project +# +# Licensed under the Apache License, Version 2.0 (the "License"); +# you may not use this file except in compliance with the License. +# You may obtain a copy of the License at +# +# http://www.apache.org/licenses/LICENSE-2.0 +# +# Unless required by applicable law or agreed to in writing, software +# distributed under the License is distributed on an "AS IS" BASIS, +# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +# See the License for the specific language governing permissions and +# limitations under the License. +# + +# If you don't need to do a full clean build but would like to touch +# a file or delete some intermediate files, add a clean step to the end +# of the list. These steps will only be run once, if they haven't been +# run before. +# +# E.g.: +# $(call add-clean-step, touch -c external/sqlite/sqlite3.h) +# $(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/STATIC_LIBRARIES/libz_intermediates) +# +# Always use "touch -c" and "rm -f" or "rm -rf" to gracefully deal with +# files that are missing or have been moved. +# +# Use $(PRODUCT_OUT) to get to the "out/target/product/blah/" directory. +# Use $(OUT_DIR) to refer to the "out" directory. +# +# If you need to re-do something that's already mentioned, just copy +# the command and add it to the bottom of the list. E.g., if a change +# that you made last week required touching a file and a change you +# made today requires touching the same file, just copy the old +# touch step and add it to the end of the list. +# +# ************************************************ +# NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST +# ************************************************ + +# For example: +#$(call add-clean-step, rm -rf $(OUT_DIR)/target/common/obj/APPS/AndroidTests_intermediates) +#$(call add-clean-step, rm -rf $(OUT_DIR)/target/common/obj/JAVA_LIBRARIES/core_intermediates) +#$(call add-clean-step, find $(OUT_DIR) -type f -name "IGTalkSession*" -print0 | xargs -0 rm -f) +#$(call add-clean-step, rm -rf $(PRODUCT_OUT)/data/*) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/SHARED_LIBRARIES/libmedia_native_intermediates) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/lib/libmedia_native.so) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/symbols/system/lib/libmedia_native.so) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libmedia_native.so) +# ************************************************ +# NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST +# ************************************************ diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp index fd91bf2..d10f2e5 100644 --- a/camera/CameraParameters.cpp +++ b/camera/CameraParameters.cpp @@ -90,6 +90,7 @@ const char CameraParameters::KEY_RECORDING_HINT[] = "recording-hint"; const char CameraParameters::KEY_VIDEO_SNAPSHOT_SUPPORTED[] = "video-snapshot-supported"; const char CameraParameters::KEY_VIDEO_STABILIZATION[] = "video-stabilization"; const char CameraParameters::KEY_VIDEO_STABILIZATION_SUPPORTED[] = "video-stabilization-supported"; +const char CameraParameters::KEY_LIGHTFX[] = "light-fx"; const char CameraParameters::TRUE[] = "true"; const char CameraParameters::FALSE[] = "false"; @@ -167,6 +168,10 @@ const char CameraParameters::FOCUS_MODE_EDOF[] = "edof"; const char CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO[] = "continuous-video"; const char CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE[] = "continuous-picture"; +// Values for light fx settings +const char CameraParameters::LIGHTFX_LOWLIGHT[] = "low-light"; +const char CameraParameters::LIGHTFX_HDR[] = "high-dynamic-range"; + CameraParameters::CameraParameters() : mMap() { diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk index 1247588..f60b1a4 100644 --- a/cmds/stagefright/Android.mk +++ b/cmds/stagefright/Android.mk @@ -8,7 +8,7 @@ LOCAL_SRC_FILES:= \ SineSource.cpp LOCAL_SHARED_LIBRARIES := \ - libstagefright libmedia libmedia_native libutils libbinder libstagefright_foundation \ + libstagefright libmedia libutils libbinder libstagefright_foundation \ libjpeg libgui LOCAL_C_INCLUDES:= \ @@ -104,7 +104,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libstagefright liblog libutils libbinder libgui \ - libstagefright_foundation libmedia libmedia_native libcutils + libstagefright_foundation libmedia libcutils LOCAL_C_INCLUDES:= \ frameworks/av/media/libstagefright \ @@ -127,7 +127,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libstagefright liblog libutils libbinder libstagefright_foundation \ - libmedia libmedia_native libgui libcutils libui + libmedia libgui libcutils libui LOCAL_C_INCLUDES:= \ frameworks/av/media/libstagefright \ @@ -151,7 +151,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libstagefright liblog libutils libbinder libstagefright_foundation \ - libmedia libmedia_native libgui libcutils libui + libmedia libgui libcutils libui LOCAL_C_INCLUDES:= \ frameworks/av/media/libstagefright \ diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp index b92a8a0..1e0e7f8 100644 --- a/cmds/stagefright/stagefright.cpp +++ b/cmds/stagefright/stagefright.cpp @@ -589,7 +589,7 @@ static void performSeekTest(const sp<MediaSource> &source) { } static void usage(const char *me) { - fprintf(stderr, "usage: %s\n", me); + fprintf(stderr, "usage: %s [options] [input_filename]\n", me); fprintf(stderr, " -h(elp)\n"); fprintf(stderr, " -a(udio)\n"); fprintf(stderr, " -n repetitions\n"); @@ -607,8 +607,8 @@ static void usage(const char *me) { "(video only)\n"); fprintf(stderr, " -S allocate buffers from a surface\n"); fprintf(stderr, " -T allocate buffers from a surface texture\n"); - fprintf(stderr, " -d(ump) filename (raw stream data to a file)\n"); - fprintf(stderr, " -D(ump) filename (decoded PCM data to a file)\n"); + fprintf(stderr, " -d(ump) output_filename (raw stream data to a file)\n"); + fprintf(stderr, " -D(ump) output_filename (decoded PCM data to a file)\n"); } static void dumpCodecProfiles(const sp<IOMX>& omx, bool queryDecoders) { diff --git a/include/camera/CameraParameters.h b/include/camera/CameraParameters.h index 5540d32..d521543 100644 --- a/include/camera/CameraParameters.h +++ b/include/camera/CameraParameters.h @@ -525,6 +525,10 @@ public: // stream and record stabilized videos. static const char KEY_VIDEO_STABILIZATION_SUPPORTED[]; + // Supported modes for special effects with light. + // Example values: "lowlight,hdr". + static const char KEY_LIGHTFX[]; + // Value for KEY_ZOOM_SUPPORTED or KEY_SMOOTH_ZOOM_SUPPORTED. static const char TRUE[]; static const char FALSE[]; @@ -664,6 +668,12 @@ public: // other modes. static const char FOCUS_MODE_CONTINUOUS_PICTURE[]; + // Values for light special effects + // Low-light enhancement mode + static const char LIGHTFX_LOWLIGHT[]; + // High-dynamic range mode + static const char LIGHTFX_HDR[]; + private: DefaultKeyedVector<String8,String8> mMap; }; diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 156c592..cd7ff92 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -43,46 +43,43 @@ public: */ enum event_type { EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer. - EVENT_OVERRUN = 1, // PCM buffer overrun occured. + EVENT_OVERRUN = 1, // PCM buffer overrun occurred. EVENT_MARKER = 2, // Record head is at the specified marker position // (See setMarkerPosition()). EVENT_NEW_POS = 3, // Record head is at a new position // (See setPositionUpdatePeriod()). }; - /* Create Buffer on the stack and pass it to obtainBuffer() - * and releaseBuffer(). + /* Client should declare Buffer on the stack and pass address to obtainBuffer() + * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. */ class Buffer { public: - enum { - MUTE = 0x00000001 - }; - uint32_t flags; - int channelCount; - audio_format_t format; - size_t frameCount; + size_t frameCount; // number of sample frames corresponding to size; + // on input it is the number of frames available, + // on output is the number of frames actually drained + size_t size; // total size in bytes == frameCount * frameSize union { void* raw; - short* i16; - int8_t* i8; + short* i16; // signed 16-bit + int8_t* i8; // unsigned 8-bit, offset by 0x80 }; }; /* As a convenience, if a callback is supplied, a handler thread * is automatically created with the appropriate priority. This thread - * invokes the callback when a new buffer becomes ready or an overrun condition occurs. + * invokes the callback when a new buffer becomes ready or various conditions occur. * Parameters: * * event: type of event notified (see enum AudioRecord::event_type). * user: Pointer to context for use by the callback receiver. * info: Pointer to optional parameter according to event type: * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read - * more bytes than indicated by 'size' field and update 'size' if less bytes are - * read. + * more bytes than indicated by 'size' field and update 'size' if fewer bytes are + * consumed. * - EVENT_OVERRUN: unused. * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. @@ -98,7 +95,7 @@ public: * - BAD_VALUE: unsupported configuration */ - static status_t getMinFrameCount(int* frameCount, + static status_t getMinFrameCount(size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask); @@ -108,7 +105,7 @@ public: */ AudioRecord(); - /* Creates an AudioRecord track and registers it with AudioFlinger. + /* Creates an AudioRecord object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to the audio hardware's current * values. @@ -120,10 +117,13 @@ public: * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). * channelMask: Channel mask. - * frameCount: Total size of track PCM buffer in frames. This defines the - * latency of the track. + * frameCount: Minimum size of track PCM buffer in frames. This defines the + * application's contribution to the + * latency of the track. The actual size selected by the AudioRecord could + * be larger if the requested size is not compatible with current audio HAL + * latency. Zero means to use a default value. * cbf: Callback function. If not null, this function is called periodically - * to provide new PCM data. + * to consume new PCM data. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames are ready in record track output buffer. @@ -154,7 +154,7 @@ public: * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process - * */ + */ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT, uint32_t sampleRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, @@ -168,14 +168,14 @@ public: /* Result of constructing the AudioRecord. This must be checked - * before using any AudioRecord API (except for set()), using + * before using any AudioRecord API (except for set()), because using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. */ status_t initCheck() const; - /* Returns this track's latency in milliseconds. - * This includes the latency due to AudioRecord buffer size + /* Returns this track's estimated latency in milliseconds. + * This includes the latency due to AudioRecord buffer size, * and audio hardware driver. */ uint32_t latency() const; @@ -184,14 +184,14 @@ public: audio_format_t format() const; int channelCount() const; - uint32_t frameCount() const; - size_t frameSize() const; + size_t frameCount() const; + size_t frameSize() const { return mFrameSize; } audio_source_t inputSource() const; /* After it's created the track is not active. Call start() to * make it active. If set, the callback will start being called. - * if event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until + * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until * the specified event occurs on the specified trigger session. */ status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, @@ -199,12 +199,12 @@ public: /* Stop a track. If set, the callback will cease being called and * obtainBuffer returns STOPPED. Note that obtainBuffer() still works - * and will fill up buffers until the pool is exhausted. + * and will drain buffers until the pool is exhausted. */ void stop(); bool stopped() const; - /* get sample rate for this record track + /* Get sample rate for this record track in Hz. */ uint32_t getSampleRate() const; @@ -258,7 +258,7 @@ public: */ status_t getPosition(uint32_t *position) const; - /* returns a handle on the audio input used by this AudioRecord. + /* Returns a handle on the audio input used by this AudioRecord. * * Parameters: * none. @@ -268,7 +268,7 @@ public: */ audio_io_handle_t getInput() const; - /* returns the audio session ID associated with this AudioRecord. + /* Returns the audio session ID associated with this AudioRecord. * * Parameters: * none. @@ -278,22 +278,30 @@ public: */ int getSessionId() const; - /* obtains a buffer of "frameCount" frames. The buffer must be - * filled entirely. If the track is stopped, obtainBuffer() returns + /* Obtains a buffer of "frameCount" frames. The buffer must be + * drained entirely, and then released with releaseBuffer(). + * If the track is stopped, obtainBuffer() returns * STOPPED instead of NO_ERROR as long as there are buffers available, * at which point NO_MORE_BUFFERS is returned. - * Buffers will be returned until the pool (buffercount()) + * Buffers will be returned until the pool * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "blocking" * parameter. + * + * Interpretation of waitCount: + * +n limits wait time to n * WAIT_PERIOD_MS, + * -1 causes an (almost) infinite wait time, + * 0 non-blocking. */ enum { - NO_MORE_BUFFERS = 0x80000001, + NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value STOPPED = 1 }; status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); + + /* Release an emptied buffer of "frameCount" frames for AudioFlinger to re-fill. */ void releaseBuffer(Buffer* audioBuffer); @@ -302,16 +310,16 @@ public: */ ssize_t read(void* buffer, size_t size); - /* Return the amount of input frames lost in the audio driver since the last call of this + /* Return the number of input frames lost in the audio driver since the last call of this * function. Audio driver is expected to reset the value to 0 and restart counting upon * returning the current value by this function call. Such loss typically occurs when the * user space process is blocked longer than the capacity of audio driver buffers. - * Unit: the number of input audio frames + * Units: the number of input audio frames. */ unsigned int getInputFramesLost() const; private: - /* copying audio tracks is not allowed */ + /* copying audio record objects is not allowed */ AudioRecord(const AudioRecord& other); AudioRecord& operator = (const AudioRecord& other); @@ -344,7 +352,7 @@ private: status_t openRecord_l(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_io_handle_t input); audio_io_handle_t getInput_l(); status_t restoreRecord_l(audio_track_cblk_t*& cblk); @@ -355,7 +363,7 @@ private: bool mActive; // protected by mLock // for client callback handler - callback_t mCbf; + callback_t mCbf; // callback handler for events, or NULL void* mUserData; // for notification APIs @@ -367,9 +375,10 @@ private: uint32_t mUpdatePeriod; // in ms // constant after constructor or set() - uint32_t mFrameCount; + size_t mFrameCount; audio_format_t mFormat; uint8_t mChannelCount; + size_t mFrameSize; // app-level frame size == AudioFlinger frame size audio_source_t mInputSource; status_t mStatus; uint32_t mLatency; @@ -381,6 +390,7 @@ private: sp<IAudioRecord> mAudioRecord; sp<IMemory> mCblkMemory; audio_track_cblk_t* mCblk; + void* mBuffers; // starting address of buffers in shared memory int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 49e1afc..126ef12 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -87,29 +87,26 @@ public: static float linearToLog(int volume); static int logToLinear(float volume); - static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); - static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); - static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputSamplingRate(uint32_t* samplingRate, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputFrameCount(size_t* frameCount, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputLatency(uint32_t* latency, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); static status_t getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, - int* samplingRate); + uint32_t* samplingRate); // returns the number of frames per audio HAL write buffer. Corresponds to // audio_stream->get_buffer_size()/audio_stream_frame_size() static status_t getFrameCount(audio_io_handle_t output, audio_stream_type_t stream, - int* frameCount); + size_t* frameCount); // returns the audio output stream latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, audio_stream_type_t stream, uint32_t* latency); - // DEPRECATED - static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT); - - // DEPRECATED - static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT); - static bool routedToA2dpOutput(audio_stream_type_t streamType); static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, @@ -126,10 +123,11 @@ public: // - BAD_VALUE: invalid parameter // NOTE: this feature is not supported on all hardware platforms and it is // necessary to check returned status before using the returned values. - static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid - static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); + static size_t getInputFramesLost(audio_io_handle_t ioHandle); static int newAudioSessionId(); static void acquireAudioSessionId(int audioSession); @@ -147,8 +145,8 @@ public: NUM_CONFIG_EVENTS }; - // audio output descriptor used to cache output configurations in client process to avoid frequent calls - // through IAudioFlinger + // audio output descriptor used to cache output configurations in client process to avoid + // frequent calls through IAudioFlinger class OutputDescriptor { public: OutputDescriptor() @@ -162,8 +160,8 @@ public: }; // Events used to synchronize actions between audio sessions. - // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until playback - // is complete on another audio session. + // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until + // playback is complete on another audio session. // See definitions in MediaSyncEvent.java enum sync_event_t { SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event @@ -183,8 +181,10 @@ public: // // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // - static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address); - static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); + static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, + const char *device_address); + static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, + const char *device_address); static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); @@ -237,8 +237,8 @@ public: static const sp<IAudioPolicyService>& get_audio_policy_service(); // helpers for android.media.AudioManager.getProperty(), see description there for meaning - static int32_t getPrimaryOutputSamplingRate(); - static int32_t getPrimaryOutputFrameCount(); + static uint32_t getPrimaryOutputSamplingRate(); + static size_t getPrimaryOutputFrameCount(); // ---------------------------------------------------------------------------- diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 34108b3..f1b26b5 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -49,13 +49,17 @@ public: }; /* Events used by AudioTrack callback function (audio_track_cblk_t). + * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. */ enum event_type { EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer. - EVENT_UNDERRUN = 1, // PCM buffer underrun occured. - EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0. - EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()). - EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()). + EVENT_UNDERRUN = 1, // PCM buffer underrun occurred. + EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from + // loop start if loop count was not 0. + EVENT_MARKER = 3, // Playback head is at the specified marker position + // (See setMarkerPosition()). + EVENT_NEW_POS = 4, // Playback head is at a new position + // (See setPositionUpdatePeriod()). EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. }; @@ -66,14 +70,6 @@ public: class Buffer { public: - enum { - MUTE = 0x00000001 - }; - uint32_t flags; // 0 or MUTE - audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT - // accessed directly by WebKit ANP callback - int channelCount; // will be removed in the future, do not use - size_t frameCount; // number of sample frames corresponding to size; // on input it is the number of frames desired, // on output is the number of frames actually filled @@ -114,7 +110,7 @@ public: * - NO_INIT: audio server or audio hardware not initialized */ - static status_t getMinFrameCount(int* frameCount, + static status_t getMinFrameCount(size_t* frameCount, audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, uint32_t sampleRate = 0); @@ -123,7 +119,7 @@ public: */ AudioTrack(); - /* Creates an audio track and registers it with AudioFlinger. + /* Creates an AudioTrack object and registers it with AudioFlinger. * Once created, the track needs to be started before it can be used. * Unspecified values are set to the audio hardware's current * values. @@ -137,12 +133,13 @@ public: * 16 bits per sample). * channelMask: Channel mask. * frameCount: Minimum size of track PCM buffer in frames. This defines the + * application's contribution to the * latency of the track. The actual size selected by the AudioTrack could be * larger if the requested size is not compatible with current audio HAL * latency. Zero means to use a default value. * flags: See comments on audio_output_flags_t in <system/audio.h>. * cbf: Callback function. If not null, this function is called periodically - * to request new PCM data. + * to provide new PCM data. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames have been consumed from track input buffer. @@ -162,18 +159,6 @@ public: int notificationFrames = 0, int sessionId = 0); - // DEPRECATED - explicit AudioTrack( int streamType, - uint32_t sampleRate = 0, - int format = AUDIO_FORMAT_DEFAULT, - int channelMask = 0, - int frameCount = 0, - uint32_t flags = (uint32_t) AUDIO_OUTPUT_FLAG_NONE, - callback_t cbf = 0, - void* user = 0, - int notificationFrames = 0, - int sessionId = 0); - /* Creates an audio track and registers it with AudioFlinger. With this constructor, * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer * identified by the argument sharedBuffer. This prototype is for static buffer playback. @@ -206,7 +191,7 @@ public: * - INVALID_OPERATION: AudioTrack is already initialized * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized - * */ + */ status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, uint32_t sampleRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, @@ -244,7 +229,7 @@ public: /* Return channelCount * (bit depth per channel / 8). * channelCount is determined from channelMask, and bit depth comes from format. */ - size_t frameSize() const; + size_t frameSize() const { return mFrameSize; } sp<IMemory>& sharedBuffer(); @@ -280,9 +265,14 @@ public: /* Set volume for this track, mostly used for games' sound effects * left and right volumes. Levels must be >= 0.0 and <= 1.0. + * This is the older API. New applications should use setVolume(float) when possible. */ status_t setVolume(float left, float right); - void getVolume(float* left, float* right) const; + + /* Set volume for all channels. This is the preferred API for new applications, + * especially for multi-channel content. + */ + status_t setVolume(float volume); /* Set the send level for this track. An auxiliary effect should be attached * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. @@ -290,9 +280,11 @@ public: status_t setAuxEffectSendLevel(float level); void getAuxEffectSendLevel(float* level) const; - /* Set sample rate for this track, mostly used for games' sound effects + /* Set sample rate for this track in Hz, mostly used for games' sound effects */ - status_t setSampleRate(int sampleRate); + status_t setSampleRate(uint32_t sampleRate); + + /* Return current sample rate in Hz, or 0 if unknown */ uint32_t getSampleRate() const; /* Enables looping and sets the start and end points of looping. @@ -312,7 +304,8 @@ public: /* Sets marker position. When playback reaches the number of frames specified, a callback with * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker * notification callback. - * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * If the AudioTrack has been opened with no callback function associated, the operation will + * fail. * * Parameters: * @@ -330,7 +323,8 @@ public: * a callback with event type EVENT_NEW_POS is called. * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification * callback. - * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * If the AudioTrack has been opened with no callback function associated, the operation will + * fail. * * Parameters: * @@ -359,7 +353,8 @@ public: * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - INVALID_OPERATION: the AudioTrack is not stopped. - * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer + * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack + * buffer */ status_t setPosition(uint32_t position); status_t getPosition(uint32_t *position); @@ -413,7 +408,7 @@ public: * If the track is stopped, obtainBuffer() returns * STOPPED instead of NO_ERROR as long as there are buffers available, * at which point NO_MORE_BUFFERS is returned. - * Buffers will be returned until the pool (buffercount()) + * Buffers will be returned until the pool * is exhausted, at which point obtainBuffer() will either block * or return WOULD_BLOCK depending on the value of the "blocking" * parameter. @@ -422,6 +417,18 @@ public: * +n limits wait time to n * WAIT_PERIOD_MS, * -1 causes an (almost) infinite wait time, * 0 non-blocking. + * + * Buffer fields + * On entry: + * frameCount number of frames requested + * After error return: + * frameCount 0 + * size 0 + * raw undefined + * After successful return: + * frameCount actual number of frames available, <= number requested + * size actual number of bytes available + * raw pointer to the buffer */ enum { @@ -482,11 +489,12 @@ protected: // body of AudioTrackThread::threadLoop() bool processAudioBuffer(const sp<AudioTrackThread>& thread); + // caller must hold lock on mLock for all _l methods status_t createTrack_l(audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output); @@ -504,35 +512,59 @@ protected: float mSendLevel; uint32_t mFrameCount; - audio_track_cblk_t* mCblk; - audio_format_t mFormat; + audio_track_cblk_t* mCblk; // re-load after mLock.unlock() + + // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers + // is the value of pointer() for the shared buffer, otherwise mBuffers points + // immediately after the control block. This address is for the mapping within client + // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. + void* mBuffers; + + audio_format_t mFormat; // as requested by client, not forced to 16-bit audio_stream_type_t mStreamType; uint8_t mChannelCount; uint8_t mMuted; uint8_t mReserved; audio_channel_mask_t mChannelMask; + + // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. + // For 8-bit PCM data, mFrameSizeAF is + // twice as large because data is expanded to 16-bit before being stored in buffer. + size_t mFrameSize; // app-level frame size + size_t mFrameSizeAF; // AudioFlinger frame size + status_t mStatus; uint32_t mLatency; bool mActive; // protected by mLock callback_t mCbf; // callback handler for events, or NULL - void* mUserData; - uint32_t mNotificationFramesReq; // requested number of frames between each notification callback - uint32_t mNotificationFramesAct; // actual number of frames between each notification callback + void* mUserData; // for client callback handler + + // for notification APIs + uint32_t mNotificationFramesReq; // requested number of frames between each + // notification callback + uint32_t mNotificationFramesAct; // actual number of frames between each + // notification callback sp<IMemory> mSharedBuffer; int mLoopCount; uint32_t mRemainingFrames; - uint32_t mMarkerPosition; + uint32_t mMarkerPosition; // in frames bool mMarkerReached; - uint32_t mNewPosition; - uint32_t mUpdatePeriod; + uint32_t mNewPosition; // in frames + uint32_t mUpdatePeriod; // in frames + bool mFlushed; // FIXME will be made obsolete by making flush() synchronous audio_output_flags_t mFlags; int mSessionId; int mAuxEffectId; + + // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: + // 1. mLock + // 2. mCblk->lock + // It is OK to lock only mCblk->lock. mutable Mutex mLock; - status_t mRestoreStatus; + bool mIsTimed; int mPreviousPriority; // before start() SchedPolicy mPreviousSchedulingGroup; diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h index 65c26f4..b1ed7b0 100644 --- a/include/media/EffectsFactoryApi.h +++ b/include/media/EffectsFactoryApi.h @@ -74,7 +74,8 @@ int EffectQueryNumberEffects(uint32_t *pNumEffects); // -ENOENT no more effect available // -ENODEV factory failed to initialize // -EINVAL invalid pDescriptor -// -ENOSYS effect list has changed since last execution of EffectQueryNumberEffects() +// -ENOSYS effect list has changed since last execution of +// EffectQueryNumberEffects() // *pDescriptor: updated with the effect descriptor. // //////////////////////////////////////////////////////////////////////////////// @@ -91,12 +92,12 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor); // // Input: // pEffectUuid: pointer to the effect uuid. -// sessionId: audio session to which this effect instance will be attached. All effects created -// with the same session ID are connected in series and process the same signal stream. -// Knowing that two effects are part of the same effect chain can help the library implement -// some kind of optimizations. -// ioId: identifies the output or input stream this effect is directed to at audio HAL. For future -// use especially with tunneled HW accelerated effects +// sessionId: audio session to which this effect instance will be attached. All effects +// created with the same session ID are connected in series and process the same signal +// stream. Knowing that two effects are part of the same effect chain can help the +// library implement some kind of optimizations. +// ioId: identifies the output or input stream this effect is directed to at audio HAL. +// For future use especially with tunneled HW accelerated effects // // Input/Output: // pHandle: address where to return the effect handle. @@ -109,7 +110,8 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor); // *pHandle: updated with the effect handle. // //////////////////////////////////////////////////////////////////////////////// -int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle); +int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, + effect_handle_t *pHandle); //////////////////////////////////////////////////////////////////////////////// // diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 5170a87..9727143 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -61,8 +61,8 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, - track_flags_t flags, + size_t frameCount, + track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, // -1 means unused, otherwise must be valid non-0 @@ -75,7 +75,7 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, track_flags_t flags, pid_t tid, // -1 means unused, otherwise must be valid non-0 int *sessionId, @@ -123,7 +123,8 @@ public: virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) = 0; - virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0; + virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) + const = 0; // register a current process for audio output change notifications virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0; @@ -156,10 +157,10 @@ public: virtual status_t setVoiceVolume(float volume) = 0; - virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, + virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, audio_io_handle_t output) const = 0; - virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0; + virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; virtual int newAudioSessionId() = 0; @@ -191,8 +192,8 @@ public: // helpers for android.media.AudioManager.getProperty(), see description there for meaning // FIXME move these APIs to AudioPolicy to permit a more accurate implementation // that looks on primary device for a stream with fast flag, primary flag, or first one. - virtual int32_t getPrimaryOutputSamplingRate() = 0; - virtual int32_t getPrimaryOutputFrameCount() = 0; + virtual uint32_t getPrimaryOutputSamplingRate() = 0; + virtual size_t getPrimaryOutputFrameCount() = 0; }; diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index cc2e069..f5b0604 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -44,9 +44,10 @@ public: audio_policy_dev_state_t state, const char *device_address) = 0; virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, - const char *device_address) = 0; + const char *device_address) = 0; virtual status_t setPhoneState(audio_mode_t state) = 0; - virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0; + virtual status_t setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) = 0; virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0; virtual audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h index 002b045..7bf3069 100644 --- a/include/media/SoundPool.h +++ b/include/media/SoundPool.h @@ -65,8 +65,10 @@ public: sp<IMemory> getIMemory() { return mData; } // hack - void init(int numChannels, int sampleRate, audio_format_t format, size_t size, sp<IMemory> data ) { - mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; mData = data; } + void init(int numChannels, int sampleRate, audio_format_t format, size_t size, + sp<IMemory> data ) { + mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; + mData = data; } private: void init(); diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h index 29c8fd9..0529bcd 100644 --- a/include/media/ToneGenerator.h +++ b/include/media/ToneGenerator.h @@ -263,7 +263,7 @@ private: unsigned short mLoopCounter; // Current tone loopback count - int mSamplingRate; // AudioFlinger Sampling rate + uint32_t mSamplingRate; // AudioFlinger Sampling rate AudioTrack *mpAudioTrack; // Pointer to audio track used for playback Mutex mLock; // Mutex to control concurent access to ToneGenerator object from audio callback and application API Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h index 81f42ed..f5d6eb5 100644 --- a/include/media/nbaio/NBAIO.h +++ b/include/media/nbaio/NBAIO.h @@ -45,17 +45,15 @@ enum { // Negotiation of format is based on the data provider and data sink, or the data consumer and // data source, exchanging prioritized arrays of offers and counter-offers until a single offer is // mutually agreed upon. Each offer is an NBAIO_Format. For simplicity and performance, -// NBAIO_Format is an enum that ties together the most important combinations of the various +// NBAIO_Format is a typedef that ties together the most important combinations of the various // attributes, rather than a struct with separate fields for format, sample rate, channel count, // interleave, packing, alignment, etc. The reason is that NBAIO_Format tries to abstract out only -// the combinations that are actually needed within AudioFligner. If the list of combinations grows +// the combinations that are actually needed within AudioFlinger. If the list of combinations grows // too large, then this decision should be re-visited. -enum NBAIO_Format { - Format_Invalid, - Format_SR44_1_C2_I16, // 44.1 kHz PCM stereo interleaved 16-bit signed - Format_SR48_C2_I16, // 48 kHz PCM stereo interleaved 16-bit signed - Format_SR44_1_C1_I16, // 44.1 kHz PCM mono interleaved 16-bit signed - Format_SR48_C1_I16, // 48 kHz PCM mono interleaved 16-bit signed +// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed. +typedef unsigned NBAIO_Format; +enum { + Format_Invalid }; // Return the frame size of an NBAIO_Format in bytes diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h index 457d5d7..81de6e4 100644 --- a/include/media/stagefright/MediaDefs.h +++ b/include/media/stagefright/MediaDefs.h @@ -42,6 +42,7 @@ extern const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW; extern const char *MEDIA_MIMETYPE_AUDIO_RAW; extern const char *MEDIA_MIMETYPE_AUDIO_FLAC; extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS; +extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM; extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4; extern const char *MEDIA_MIMETYPE_CONTAINER_WAV; diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 5b133f3..bbc5e26 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -27,33 +27,16 @@ namespace android { // ---------------------------------------------------------------------------- // Maximum cumulated timeout milliseconds before restarting audioflinger thread -#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP init time +#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP + // init time #define MAX_RUN_TIMEOUT_MS 1000 #define WAIT_PERIOD_MS 10 -#define RESTORE_TIMEOUT_MS 5000 // Maximum waiting time for a track to be restored - -#define CBLK_UNDERRUN_MSK 0x0001 -#define CBLK_UNDERRUN_ON 0x0001 // underrun (out) or overrrun (in) indication -#define CBLK_UNDERRUN_OFF 0x0000 // no underrun -#define CBLK_DIRECTION_MSK 0x0002 -#define CBLK_DIRECTION_OUT 0x0002 // this cblk is for an AudioTrack -#define CBLK_DIRECTION_IN 0x0000 // this cblk is for an AudioRecord -#define CBLK_FORCEREADY_MSK 0x0004 -#define CBLK_FORCEREADY_ON 0x0004 // track is considered ready immediately by AudioFlinger -#define CBLK_FORCEREADY_OFF 0x0000 // track is ready when buffer full -#define CBLK_INVALID_MSK 0x0008 -#define CBLK_INVALID_ON 0x0008 // track buffer is invalidated by AudioFlinger: -#define CBLK_INVALID_OFF 0x0000 // must be re-created -#define CBLK_DISABLED_MSK 0x0010 -#define CBLK_DISABLED_ON 0x0010 // track disabled by AudioFlinger due to underrun: -#define CBLK_DISABLED_OFF 0x0000 // must be re-started -#define CBLK_RESTORING_MSK 0x0020 -#define CBLK_RESTORING_ON 0x0020 // track is being restored after invalidation -#define CBLK_RESTORING_OFF 0x0000 // by AudioFlinger -#define CBLK_RESTORED_MSK 0x0040 -#define CBLK_RESTORED_ON 0x0040 // track has been restored after invalidation -#define CBLK_RESTORED_OFF 0x0040 // by AudioFlinger -#define CBLK_FAST 0x0080 // AudioFlinger successfully created a fast track + +#define CBLK_UNDERRUN 0x01 // set: underrun (out) or overrrun (in), clear: no underrun or overrun +#define CBLK_FORCEREADY 0x02 // set: track is considered ready immediately by AudioFlinger, + // clear: track is ready when buffer full +#define CBLK_INVALID 0x04 // track buffer invalidated by AudioFlinger, need to re-create +#define CBLK_DISABLED 0x08 // track disabled by AudioFlinger due to underrun, need to re-start // Important: do not add any virtual methods, including ~ struct audio_track_cblk_t @@ -70,12 +53,11 @@ struct audio_track_cblk_t uint32_t userBase; uint32_t serverBase; - // if there is a shared buffer, "buffers" is the value of pointer() for the shared - // buffer, otherwise "buffers" points immediately after the control block - void* buffers; + int mPad1; // unused, but preserves cache line alignment + uint32_t frameCount; - // Cache line boundary + // Cache line boundary (32 bytes) uint32_t loopStart; uint32_t loopEnd; // read-only for server, read/write for client @@ -91,16 +73,14 @@ public: uint32_t sampleRate; - // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for - // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of - // 16 bit because data is converted to 16 bit before being stored in buffer + uint8_t mPad2; // unused // read-only for client, server writes once at initialization and is then read-only - uint8_t frameSize; // would normally be size_t, but 8 bits is plenty uint8_t mName; // normal tracks: track name, fast tracks: track index // used by client only - uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger + uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting + // audioflinger uint16_t waitTimeMs; // Cumulated wait time, used by client only private: @@ -114,13 +94,25 @@ public: // Since the control block is always located in shared memory, this constructor // is only used for placement new(). It is never used for regular new() or stack. audio_track_cblk_t(); - uint32_t stepUser(uint32_t frameCount); // called by client only, where - // client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack - bool stepServer(uint32_t frameCount); // called by server only - void* buffer(uint32_t offset) const; - uint32_t framesAvailable(); - uint32_t framesAvailable_l(); - uint32_t framesReady(); // called by server only + + // called by client only, where client includes regular + // AudioTrack and AudioFlinger::PlaybackThread::OutputTrack + uint32_t stepUserIn(uint32_t frameCount) { return stepUser(frameCount, false); } + uint32_t stepUserOut(uint32_t frameCount) { return stepUser(frameCount, true); } + + bool stepServer(uint32_t frameCount, bool isOut); + + // if there is a shared buffer, "buffers" is the value of pointer() for the shared + // buffer, otherwise "buffers" points immediately after the control block + void* buffer(void *buffers, uint32_t frameSize, uint32_t offset) const; + + uint32_t framesAvailableIn() { return framesAvailable(false); } + uint32_t framesAvailableOut() { return framesAvailable(true); } + uint32_t framesAvailableIn_l() { return framesAvailable_l(false); } + uint32_t framesAvailableOut_l() { return framesAvailable_l(true); } + uint32_t framesReadyIn() { return framesReady(false); } + uint32_t framesReadyOut() { return framesReady(true); } + bool tryLock(); // No barriers on the following operations, so the ordering of loads/stores @@ -146,6 +138,12 @@ public: return mVolumeLR; } +private: + // isOut == true means AudioTrack, isOut == false means AudioRecord + uint32_t stepUser(uint32_t frameCount, bool isOut); + uint32_t framesAvailable(bool isOut); + uint32_t framesAvailable_l(bool isOut); + uint32_t framesReady(bool isOut); }; diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk index 0ed7e6c..778c5ac 100755 --- a/libvideoeditor/lvpp/Android.mk +++ b/libvideoeditor/lvpp/Android.mk @@ -54,7 +54,6 @@ LOCAL_SHARED_LIBRARIES := \ libGLESv2 \ libgui \ libmedia \ - libmedia_native \ libdrmframework \ libstagefright \ libstagefright_foundation \ diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp index fc9fb49..a47fc15 100755 --- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp +++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp @@ -406,8 +406,8 @@ status_t VideoEditorPlayer::VeAudioOutput::open( } ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount); if (mTrack) close(); - int afSampleRate; - int afFrameCount; + uint32_t afSampleRate; + size_t afFrameCount; int frameCount; if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk index 95ca6fd..3052ad9 100644 --- a/media/libeffects/downmix/Android.mk +++ b/media/libeffects/downmix/Android.mk @@ -25,4 +25,6 @@ LOCAL_C_INCLUDES := \ LOCAL_PRELINK_MODULE := false +LOCAL_CFLAGS += -fvisibility=hidden + include $(BUILD_SHARED_LIBRARY) diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c index 5bf052a..f17a6e8 100644 --- a/media/libeffects/downmix/EffectDownmix.c +++ b/media/libeffects/downmix/EffectDownmix.c @@ -58,13 +58,13 @@ const struct effect_interface_s gDownmixInterface = { NULL /* no process_reverse function, no reference stream needed */ }; +// This is the only symbol that needs to be exported +__attribute__ ((visibility ("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { tag : AUDIO_EFFECT_LIBRARY_TAG, version : EFFECT_LIBRARY_API_VERSION, name : "Downmix Library", implementor : "The Android Open Source Project", - query_num_effects : DownmixLib_QueryNumberEffects, - query_effect : DownmixLib_QueryEffect, create_effect : DownmixLib_Create, release_effect : DownmixLib_Release, get_descriptor : DownmixLib_GetDescriptor, @@ -159,25 +159,6 @@ void Downmix_testIndexComputation(uint32_t mask) { /*--- Effect Library Interface Implementation ---*/ -int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects) { - ALOGV("DownmixLib_QueryNumberEffects()"); - *pNumEffects = kNbEffects; - return 0; -} - -int32_t DownmixLib_QueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) { - ALOGV("DownmixLib_QueryEffect() index=%d", index); - if (pDescriptor == NULL) { - return -EINVAL; - } - if (index >= (uint32_t)kNbEffects) { - return -EINVAL; - } - memcpy(pDescriptor, gDescriptors[index], sizeof(effect_descriptor_t)); - return 0; -} - - int32_t DownmixLib_Create(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, diff --git a/media/libeffects/downmix/EffectDownmix.h b/media/libeffects/downmix/EffectDownmix.h index be3ca3f..cb6b957 100644 --- a/media/libeffects/downmix/EffectDownmix.h +++ b/media/libeffects/downmix/EffectDownmix.h @@ -65,9 +65,6 @@ const uint32_t kUnsupported = * Effect API *------------------------------------ */ -int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects); -int32_t DownmixLib_QueryEffect(uint32_t index, - effect_descriptor_t *pDescriptor); int32_t DownmixLib_Create(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, diff --git a/media/libeffects/lvm/lib/Android.mk b/media/libeffects/lvm/lib/Android.mk index f49267e..bb56c75 100644 --- a/media/libeffects/lvm/lib/Android.mk +++ b/media/libeffects/lvm/lib/Android.mk @@ -105,8 +105,6 @@ LOCAL_SRC_FILES:= \ LOCAL_MODULE:= libmusicbundle - - LOCAL_C_INCLUDES += \ $(LOCAL_PATH)/Eq/lib \ $(LOCAL_PATH)/Eq/src \ @@ -121,8 +119,12 @@ LOCAL_C_INCLUDES += \ $(LOCAL_PATH)/StereoWidening/src \ $(LOCAL_PATH)/StereoWidening/lib +LOCAL_CFLAGS += -fvisibility=hidden + include $(BUILD_STATIC_LIBRARY) + + # Reverb library include $(CLEAR_VARS) @@ -168,12 +170,11 @@ LOCAL_SRC_FILES:= \ LOCAL_MODULE:= libreverb - - LOCAL_C_INCLUDES += \ $(LOCAL_PATH)/Reverb/lib \ $(LOCAL_PATH)/Reverb/src \ $(LOCAL_PATH)/Common/lib \ $(LOCAL_PATH)/Common/src +LOCAL_CFLAGS += -fvisibility=hidden include $(BUILD_STATIC_LIBRARY) diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk index 4313424..f1af389 100644 --- a/media/libeffects/lvm/wrapper/Android.mk +++ b/media/libeffects/lvm/wrapper/Android.mk @@ -9,28 +9,27 @@ LOCAL_ARM_MODE := arm LOCAL_SRC_FILES:= \ Bundle/EffectBundle.cpp +LOCAL_CFLAGS += -fvisibility=hidden + LOCAL_MODULE:= libbundlewrapper LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx - - LOCAL_STATIC_LIBRARIES += libmusicbundle LOCAL_SHARED_LIBRARIES := \ libcutils \ libdl - LOCAL_C_INCLUDES += \ $(LOCAL_PATH)/Bundle \ $(LOCAL_PATH)/../lib/Common/lib/ \ $(LOCAL_PATH)/../lib/Bundle/lib/ \ $(call include-path-for, audio-effects) - include $(BUILD_SHARED_LIBRARY) + # reverb wrapper include $(CLEAR_VARS) @@ -39,12 +38,12 @@ LOCAL_ARM_MODE := arm LOCAL_SRC_FILES:= \ Reverb/EffectReverb.cpp +LOCAL_CFLAGS += -fvisibility=hidden + LOCAL_MODULE:= libreverbwrapper LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx - - LOCAL_STATIC_LIBRARIES += libreverb LOCAL_SHARED_LIBRARIES := \ diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp index d706c2d..94b9acf 100644 --- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp +++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp @@ -158,42 +158,6 @@ int Volume_getParameter (EffectContext *pContext, int Effect_setEnabled(EffectContext *pContext, bool enabled); /* Effect Library Interface Implementation */ -extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects){ - ALOGV("\n\tEffectQueryNumberEffects start"); - *pNumEffects = 4; - ALOGV("\tEffectQueryNumberEffects creating %d effects", *pNumEffects); - ALOGV("\tEffectQueryNumberEffects end\n"); - return 0; -} /* end EffectQueryNumberEffects */ - -extern "C" int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor){ - ALOGV("\n\tEffectQueryEffect start"); - ALOGV("\tEffectQueryEffect processing index %d", index); - - if (pDescriptor == NULL){ - ALOGV("\tLVM_ERROR : EffectQueryEffect was passed NULL pointer"); - return -EINVAL; - } - if (index > 3){ - ALOGV("\tLVM_ERROR : EffectQueryEffect index out of range %d", index); - return -ENOENT; - } - if(index == LVM_BASS_BOOST){ - ALOGV("\tEffectQueryEffect processing LVM_BASS_BOOST"); - *pDescriptor = gBassBoostDescriptor; - }else if(index == LVM_VIRTUALIZER){ - ALOGV("\tEffectQueryEffect processing LVM_VIRTUALIZER"); - *pDescriptor = gVirtualizerDescriptor; - } else if(index == LVM_EQUALIZER){ - ALOGV("\tEffectQueryEffect processing LVM_EQUALIZER"); - *pDescriptor = gEqualizerDescriptor; - } else if(index == LVM_VOLUME){ - ALOGV("\tEffectQueryEffect processing LVM_VOLUME"); - *pDescriptor = gVolumeDescriptor; - } - ALOGV("\tEffectQueryEffect end\n"); - return 0; -} /* end EffectQueryEffect */ extern "C" int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, @@ -3299,13 +3263,13 @@ const struct effect_interface_s gLvmEffectInterface = { NULL, }; /* end gLvmEffectInterface */ +// This is the only symbol that needs to be exported +__attribute__ ((visibility ("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { tag : AUDIO_EFFECT_LIBRARY_TAG, version : EFFECT_LIBRARY_API_VERSION, name : "Effect Bundle Library", implementor : "NXP Software Ltd.", - query_num_effects : android::EffectQueryNumberEffects, - query_effect : android::EffectQueryEffect, create_effect : android::EffectCreate, release_effect : android::EffectRelease, get_descriptor : android::EffectGetDescriptor, diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp index 941d651..87e2c85 100755..100644 --- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp +++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp @@ -186,30 +186,6 @@ int Reverb_getParameter (ReverbContext *pContext, int Reverb_LoadPreset (ReverbContext *pContext); /* Effect Library Interface Implementation */ -extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects){ - ALOGV("\n\tEffectQueryNumberEffects start"); - *pNumEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *); - ALOGV("\tEffectQueryNumberEffects creating %d effects", *pNumEffects); - ALOGV("\tEffectQueryNumberEffects end\n"); - return 0; -} /* end EffectQueryNumberEffects */ - -extern "C" int EffectQueryEffect(uint32_t index, - effect_descriptor_t *pDescriptor){ - ALOGV("\n\tEffectQueryEffect start"); - ALOGV("\tEffectQueryEffect processing index %d", index); - if (pDescriptor == NULL){ - ALOGV("\tLVM_ERROR : EffectQueryEffect was passed NULL pointer"); - return -EINVAL; - } - if (index >= sizeof(gDescriptors) / sizeof(const effect_descriptor_t *)) { - ALOGV("\tLVM_ERROR : EffectQueryEffect index out of range %d", index); - return -ENOENT; - } - *pDescriptor = *gDescriptors[index]; - ALOGV("\tEffectQueryEffect end\n"); - return 0; -} /* end EffectQueryEffect */ extern "C" int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, @@ -2170,13 +2146,13 @@ const struct effect_interface_s gReverbInterface = { NULL, }; /* end gReverbInterface */ +// This is the only symbol that needs to be exported +__attribute__ ((visibility ("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { tag : AUDIO_EFFECT_LIBRARY_TAG, version : EFFECT_LIBRARY_API_VERSION, name : "Reverb Library", implementor : "NXP Software Ltd.", - query_num_effects : android::EffectQueryNumberEffects, - query_effect : android::EffectQueryEffect, create_effect : android::EffectCreate, release_effect : android::EffectRelease, get_descriptor : android::EffectGetDescriptor, diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk index c13b9d4..dfa1711 100755..100644 --- a/media/libeffects/preprocessing/Android.mk +++ b/media/libeffects/preprocessing/Android.mk @@ -29,4 +29,6 @@ else LOCAL_SHARED_LIBRARIES += libdl endif +LOCAL_CFLAGS += -fvisibility=hidden + include $(BUILD_SHARED_LIBRARY) diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp index 597866a..25586e8 100755..100644 --- a/media/libeffects/preprocessing/PreProcessing.cpp +++ b/media/libeffects/preprocessing/PreProcessing.cpp @@ -1818,30 +1818,6 @@ const struct effect_interface_s sEffectInterfaceReverse = { // Effect Library Interface Implementation //------------------------------------------------------------------------------ -int PreProcessingLib_QueryNumberEffects(uint32_t *pNumEffects) -{ - if (PreProc_Init() != 0) { - return sInitStatus; - } - if (pNumEffects == NULL) { - return -EINVAL; - } - *pNumEffects = PREPROC_NUM_EFFECTS; - return sInitStatus; -} - -int PreProcessingLib_QueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) -{ - if (PreProc_Init() != 0) { - return sInitStatus; - } - if (index >= PREPROC_NUM_EFFECTS) { - return -EINVAL; - } - *pDescriptor = *sDescriptors[index]; - return 0; -} - int PreProcessingLib_Create(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, @@ -1913,13 +1889,13 @@ int PreProcessingLib_GetDescriptor(const effect_uuid_t *uuid, return 0; } +// This is the only symbol that needs to be exported +__attribute__ ((visibility ("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { tag : AUDIO_EFFECT_LIBRARY_TAG, version : EFFECT_LIBRARY_API_VERSION, name : "Audio Preprocessing Library", implementor : "The Android Open Source Project", - query_num_effects : PreProcessingLib_QueryNumberEffects, - query_effect : PreProcessingLib_QueryEffect, create_effect : PreProcessingLib_Create, release_effect : PreProcessingLib_Release, get_descriptor : PreProcessingLib_GetDescriptor diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp index 90ebe1f..c35453b 100644 --- a/media/libeffects/testlibs/EffectEqualizer.cpp +++ b/media/libeffects/testlibs/EffectEqualizer.cpp @@ -123,23 +123,6 @@ int Equalizer_setParameter(AudioEqualizer * pEqualizer, int32_t *pParam, void *p //--- Effect Library Interface Implementation // -extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects) { - *pNumEffects = 1; - return 0; -} /* end EffectQueryNumberEffects */ - -extern "C" int EffectQueryEffect(uint32_t index, - effect_descriptor_t *pDescriptor) { - if (pDescriptor == NULL) { - return -EINVAL; - } - if (index > 0) { - return -EINVAL; - } - *pDescriptor = gEqualizerDescriptor; - return 0; -} /* end EffectQueryNext */ - extern "C" int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, @@ -771,8 +754,6 @@ audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { version : EFFECT_LIBRARY_API_VERSION, name : "Test Equalizer Library", implementor : "The Android Open Source Project", - query_num_effects : android::EffectQueryNumberEffects, - query_effect : android::EffectQueryEffect, create_effect : android::EffectCreate, release_effect : android::EffectRelease, get_descriptor : android::EffectGetDescriptor, diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c index a87a834..c37f392 100644 --- a/media/libeffects/testlibs/EffectReverb.c +++ b/media/libeffects/testlibs/EffectReverb.c @@ -94,23 +94,6 @@ static const effect_descriptor_t * const gDescriptors[] = { /*--- Effect Library Interface Implementation ---*/ -int EffectQueryNumberEffects(uint32_t *pNumEffects) { - *pNumEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *); - return 0; -} - -int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) { - if (pDescriptor == NULL) { - return -EINVAL; - } - if (index >= sizeof(gDescriptors) / sizeof(const effect_descriptor_t *)) { - return -EINVAL; - } - memcpy(pDescriptor, gDescriptors[index], - sizeof(effect_descriptor_t)); - return 0; -} - int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, @@ -2222,8 +2205,6 @@ audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { .version = EFFECT_LIBRARY_API_VERSION, .name = "Test Equalizer Library", .implementor = "The Android Open Source Project", - .query_num_effects = EffectQueryNumberEffects, - .query_effect = EffectQueryEffect, .create_effect = EffectCreate, .release_effect = EffectRelease, .get_descriptor = EffectGetDescriptor, diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h index 1fb14a7..e5248fe 100644 --- a/media/libeffects/testlibs/EffectReverb.h +++ b/media/libeffects/testlibs/EffectReverb.h @@ -300,9 +300,6 @@ typedef struct reverb_module_s { * Effect API *------------------------------------ */ -int EffectQueryNumberEffects(uint32_t *pNumEffects); -int EffectQueryEffect(uint32_t index, - effect_descriptor_t *pDescriptor); int EffectCreate(const effect_uuid_t *effectUID, int32_t sessionId, int32_t ioId, diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk index 76b5110..49cf4fa 100644 --- a/media/libeffects/visualizer/Android.mk +++ b/media/libeffects/visualizer/Android.mk @@ -6,7 +6,7 @@ include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ EffectVisualizer.cpp -LOCAL_CFLAGS+= -O2 +LOCAL_CFLAGS+= -O2 -fvisibility=hidden LOCAL_SHARED_LIBRARIES := \ libcutils \ diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp index 44baf93..e7eccf1 100644 --- a/media/libeffects/visualizer/EffectVisualizer.cpp +++ b/media/libeffects/visualizer/EffectVisualizer.cpp @@ -177,23 +177,6 @@ int Visualizer_init(VisualizerContext *pContext) //--- Effect Library Interface Implementation // -int VisualizerLib_QueryNumberEffects(uint32_t *pNumEffects) { - *pNumEffects = 1; - return 0; -} - -int VisualizerLib_QueryEffect(uint32_t index, - effect_descriptor_t *pDescriptor) { - if (pDescriptor == NULL) { - return -EINVAL; - } - if (index > 0) { - return -EINVAL; - } - *pDescriptor = gVisualizerDescriptor; - return 0; -} - int VisualizerLib_Create(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, @@ -574,14 +557,13 @@ const struct effect_interface_s gVisualizerInterface = { NULL, }; - +// This is the only symbol that needs to be exported +__attribute__ ((visibility ("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { tag : AUDIO_EFFECT_LIBRARY_TAG, version : EFFECT_LIBRARY_API_VERSION, name : "Visualizer Library", implementor : "The Android Open Source Project", - query_num_effects : VisualizerLib_QueryNumberEffects, - query_effect : VisualizerLib_QueryEffect, create_effect : VisualizerLib_Create, release_effect : VisualizerLib_Release, get_descriptor : VisualizerLib_GetDescriptor, diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk index 54666fb..f2b6441 100644 --- a/media/libmedia/Android.mk +++ b/media/libmedia/Android.mk @@ -54,7 +54,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libui libcutils libutils libbinder libsonivox libicuuc libexpat \ libcamera_client libstagefright_foundation \ - libgui libdl libaudioutils libmedia_native + libgui libdl libaudioutils LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp index 680604b..3317d57 100644 --- a/media/libmedia/AudioEffect.cpp +++ b/media/libmedia/AudioEffect.cpp @@ -152,7 +152,8 @@ status_t AudioEffect::set(const effect_uuid_t *type, mCblk->buffer = (uint8_t *)mCblk + bufOffset; iEffect->asBinder()->linkToDeath(mIEffectClient); - ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled); + ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, + mStatus, mEnabled); return mStatus; } @@ -266,9 +267,11 @@ status_t AudioEffect::setParameter(effect_param_t *param) uint32_t size = sizeof(int); uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; - ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1); + ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data, + (param->psize == 8) ? *((int *)param->data + 1): -1); - return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size, ¶m->status); + return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size, + ¶m->status); } status_t AudioEffect::setParameterDeferred(effect_param_t *param) @@ -321,11 +324,14 @@ status_t AudioEffect::getParameter(effect_param_t *param) return BAD_VALUE; } - ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1); + ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data, + (param->psize == 8) ? *((int *)param->data + 1): -1); - uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; + uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + + param->vsize; - return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param, &psize, param); + return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param, + &psize, param); } @@ -346,7 +352,8 @@ void AudioEffect::binderDied() void AudioEffect::controlStatusChanged(bool controlGranted) { - ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf, mUserData); + ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf, + mUserData); if (controlGranted) { if (mStatus == ALREADY_EXISTS) { mStatus = NO_ERROR; diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 8ea6306..0587651 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -36,7 +36,7 @@ namespace android { // static status_t AudioRecord::getMinFrameCount( - int* frameCount, + size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) @@ -54,7 +54,7 @@ status_t AudioRecord::getMinFrameCount( } if (size == 0) { - ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x", + ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x", sampleRate, format, channelMask); return BAD_VALUE; } @@ -119,15 +119,22 @@ status_t AudioRecord::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + int frameCountInt, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) { + // FIXME "int" here is legacy and will be replaced by size_t later + if (frameCountInt < 0) { + ALOGE("Invalid frame count %d", frameCountInt); + return BAD_VALUE; + } + size_t frameCount = frameCountInt; - ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount); + ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, + frameCount); AutoMutex lock(mLock); @@ -176,7 +183,7 @@ status_t AudioRecord::set( } // validate framecount - int minFrameCount = 0; + size_t minFrameCount = 0; status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); if (status != NO_ERROR) { return status; @@ -212,6 +219,13 @@ status_t AudioRecord::set( mFrameCount = mCblk->frameCount; mChannelCount = (uint8_t)channelCount; mChannelMask = channelMask; + + if (audio_is_linear_pcm(mFormat)) { + mFrameSize = channelCount * audio_bytes_per_sample(format); + } else { + mFrameSize = sizeof(uint8_t); + } + mActive = false; mCbf = cbf; mNotificationFrames = notificationFrames; @@ -252,20 +266,11 @@ int AudioRecord::channelCount() const return mChannelCount; } -uint32_t AudioRecord::frameCount() const +size_t AudioRecord::frameCount() const { return mFrameCount; } -size_t AudioRecord::frameSize() const -{ - if (audio_is_linear_pcm(mFormat)) { - return channelCount()*audio_bytes_per_sample(mFormat); - } else { - return sizeof(uint8_t); - } -} - audio_source_t AudioRecord::inputSource() const { return mInputSource; @@ -291,17 +296,19 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) mActive = true; cblk->lock.lock(); - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { cblk->lock.unlock(); ALOGV("mAudioRecord->start()"); ret = mAudioRecord->start(event, triggerSession); cblk->lock.lock(); if (ret == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } - if (cblk->flags & CBLK_INVALID_MSK) { - ret = restoreRecord_l(cblk); + if (cblk->flags & CBLK_INVALID) { + audio_track_cblk_t* temp = cblk; + ret = restoreRecord_l(temp); + cblk = temp; } cblk->lock.unlock(); if (ret == NO_ERROR) { @@ -426,12 +433,13 @@ status_t AudioRecord::openRecord_l( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_io_handle_t input) { status_t status; const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { + ALOGE("Could not get audioflinger"); return NO_INIT; } @@ -454,20 +462,20 @@ status_t AudioRecord::openRecord_l( ALOGE("AudioFlinger could not create record track, status: %d", status); return status; } - sp<IMemory> cblk = record->getCblk(); - if (cblk == 0) { + sp<IMemory> iMem = record->getCblk(); + if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } mAudioRecord.clear(); mAudioRecord = record; mCblkMemory.clear(); - mCblkMemory = cblk; - mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); - android_atomic_and(~CBLK_DIRECTION_MSK, &mCblk->flags); - mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - mCblk->waitTimeMs = 0; + mCblkMemory = iMem; + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); + mCblk = cblk; + mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); + cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + cblk->waitTimeMs = 0; return NO_ERROR; } @@ -483,7 +491,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) audioBuffer->frameCount = 0; audioBuffer->size = 0; - uint32_t framesReady = cblk->framesReady(); + uint32_t framesReady = cblk->framesReadyIn(); if (framesReady == 0) { cblk->lock.lock(); @@ -498,17 +506,22 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) cblk->lock.unlock(); return WOULD_BLOCK; } - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { mLock.unlock(); + // this condition is in shared memory, so if IAudioRecord and control block + // are replaced due to mediaserver death or IAudioRecord invalidation then + // cv won't be signalled, but fortunately the timeout will limit the wait result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); cblk->lock.unlock(); mLock.lock(); if (!mActive) { return status_t(STOPPED); } + // IAudioRecord may have been re-created while mLock was unlocked + cblk = mCblk; cblk->lock.lock(); } - if (cblk->flags & CBLK_INVALID_MSK) { + if (cblk->flags & CBLK_INVALID) { goto create_new_record; } if (CC_UNLIKELY(result != NO_ERROR)) { @@ -521,9 +534,11 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); cblk->lock.lock(); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); create_new_record: - result = AudioRecord::restoreRecord_l(cblk); + audio_track_cblk_t* temp = cblk; + result = AudioRecord::restoreRecord_l(temp); + cblk = temp; } if (result != NO_ERROR) { ALOGW("obtainBuffer create Track error %d", result); @@ -539,7 +554,7 @@ create_new_record: } // read the server count again start_loop_here: - framesReady = cblk->framesReady(); + framesReady = cblk->framesReadyIn(); } cblk->lock.unlock(); } @@ -559,12 +574,9 @@ create_new_record: framesReq = bufferEnd - u; } - audioBuffer->flags = 0; - audioBuffer->channelCount= mChannelCount; - audioBuffer->format = mFormat; audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq*cblk->frameSize; - audioBuffer->raw = (int8_t*)cblk->buffer(u); + audioBuffer->size = framesReq * mFrameSize; + audioBuffer->raw = cblk->buffer(mBuffers, mFrameSize, u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } @@ -572,7 +584,7 @@ create_new_record: void AudioRecord::releaseBuffer(Buffer* audioBuffer) { AutoMutex lock(mLock); - mCblk->stepUser(audioBuffer->frameCount); + mCblk->stepUserIn(audioBuffer->frameCount); } audio_io_handle_t AudioRecord::getInput() const @@ -701,7 +713,8 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) status_t err = obtainBuffer(&audioBuffer, 1); if (err < NO_ERROR) { if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); + ALOGE_IF(err != status_t(NO_MORE_BUFFERS), + "Error obtaining an audio buffer, giving up."); return false; } break; @@ -733,11 +746,11 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) // Manage overrun callback - if (active && (cblk->framesAvailable() == 0)) { + if (active && (cblk->framesAvailableIn() == 0)) { // The value of active is stale, but we are almost sure to be active here because // otherwise we would have exited when obtainBuffer returned STOPPED earlier. ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { + if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { mCbf(EVENT_OVERRUN, mUserData, NULL); } } @@ -753,57 +766,41 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) // must be called with mLock and cblk.lock held. Callers must also hold strong references on // the IAudioRecord and IMemory in case they are recreated here. // If the IAudioRecord is successfully restored, the cblk pointer is updated -status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& cblk) +status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk) { status_t result; - if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { - ALOGW("dead IAudioRecord, creating a new one"); - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); + audio_track_cblk_t* cblk = refCblk; + audio_track_cblk_t* newCblk = cblk; + ALOGW("dead IAudioRecord, creating a new one"); - // if the new IAudioRecord is created, openRecord_l() will modify the - // following member variables: mAudioRecord, mCblkMemory and mCblk. - // It will also delete the strong references on previous IAudioRecord and IMemory - result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask, - mFrameCount, getInput_l()); - if (result == NO_ERROR) { - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); - } - if (result != NO_ERROR) { - mActive = false; - } + // signal old cblk condition so that other threads waiting for available buffers stop + // waiting now + cblk->cv.broadcast(); + cblk->lock.unlock(); - // signal old cblk condition for other threads waiting for restore completion - android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); - cblk->cv.broadcast(); - } else { - if (!(cblk->flags & CBLK_RESTORED_MSK)) { - ALOGW("dead IAudioRecord, waiting for a new one to be created"); - mLock.unlock(); - result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); - cblk->lock.unlock(); - mLock.lock(); - } else { - ALOGW("dead IAudioRecord, already restored"); - result = NO_ERROR; - cblk->lock.unlock(); - } - if (result != NO_ERROR || !mActive) { - result = status_t(STOPPED); - } + // if the new IAudioRecord is created, openRecord_l() will modify the + // following member variables: mAudioRecord, mCblkMemory and mCblk. + // It will also delete the strong references on previous IAudioRecord and IMemory + result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask, + mFrameCount, getInput_l()); + if (result == NO_ERROR) { + newCblk = mCblk; + // callback thread or sync event hasn't changed + result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); } + if (result != NO_ERROR) { + mActive = false; + } + ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); + result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); if (result == NO_ERROR) { // from now on we switch to the newly created cblk - cblk = mCblk; + refCblk = newCblk; } - cblk->lock.lock(); + newCblk->lock.lock(); ALOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result); diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 207f96f..028e4a3 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -205,12 +205,7 @@ int AudioSystem::logToLinear(float volume) return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0; } -// DEPRECATED -status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) { - return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType); -} - -status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType) +status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType) { audio_io_handle_t output; @@ -228,7 +223,7 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type status_t AudioSystem::getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, - int* samplingRate) + uint32_t* samplingRate) { OutputDescriptor *outputDesc; @@ -246,17 +241,13 @@ status_t AudioSystem::getSamplingRate(audio_io_handle_t output, gLock.unlock(); } - ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate); + ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output, + *samplingRate); return NO_ERROR; } -// DEPRECATED -status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) { - return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType); -} - -status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType) +status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_t streamType) { audio_io_handle_t output; @@ -274,7 +265,7 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t s status_t AudioSystem::getFrameCount(audio_io_handle_t output, audio_stream_type_t streamType, - int* frameCount) + size_t* frameCount) { OutputDescriptor *outputDesc; @@ -290,7 +281,8 @@ status_t AudioSystem::getFrameCount(audio_io_handle_t output, gLock.unlock(); } - ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output, *frameCount); + ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output, + *frameCount); return NO_ERROR; } @@ -369,7 +361,8 @@ status_t AudioSystem::setVoiceVolume(float value) return af->setVoiceVolume(value); } -status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream) +status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames, + audio_stream_type_t stream) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; @@ -381,7 +374,7 @@ status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames return af->getRenderPosition(halFrames, dspFrames, getOutput(stream)); } -unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { +size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); unsigned int result = 0; if (af == 0) return result; @@ -449,8 +442,10 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle OutputDescriptor *outputDesc = new OutputDescriptor(*desc); gOutputs.add(ioHandle, outputDesc); - ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d latency %d", - outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency); + ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %u " + "latency %d", + outputDesc->samplingRate, outputDesc->format, outputDesc->channels, + outputDesc->frameCount, outputDesc->latency); } break; case OUTPUT_CLOSED: { if (gOutputs.indexOfKey(ioHandle) < 0) { @@ -471,7 +466,8 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle if (param2 == NULL) break; desc = (const OutputDescriptor *)param2; - ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x frameCount %d latency %d", + ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x " + "frameCount %d latency %d", ioHandle, desc->samplingRate, desc->format, desc->channels, desc->frameCount, desc->latency); OutputDescriptor *outputDesc = gOutputs.valueAt(index); @@ -510,7 +506,7 @@ sp<IAudioPolicyService> AudioSystem::gAudioPolicyService; sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient; -// establish binder interface to AudioFlinger service +// establish binder interface to AudioPolicy service const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service() { gLock.lock(); @@ -744,14 +740,14 @@ status_t AudioSystem::isSourceActive(audio_source_t stream, bool* state) return NO_ERROR; } -int32_t AudioSystem::getPrimaryOutputSamplingRate() +uint32_t AudioSystem::getPrimaryOutputSamplingRate() { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return 0; return af->getPrimaryOutputSamplingRate(); } -int32_t AudioSystem::getPrimaryOutputFrameCount() +size_t AudioSystem::getPrimaryOutputFrameCount() { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return 0; diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 362d022..979ee37 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -50,7 +50,7 @@ namespace android { // static status_t AudioTrack::getMinFrameCount( - int* frameCount, + size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) { @@ -65,11 +65,11 @@ status_t AudioTrack::getMinFrameCount( // audio_format_t format // audio_channel_mask_t channelMask // audio_output_flags_t flags - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { return NO_INIT; } - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { return NO_INIT; } @@ -120,28 +120,6 @@ AudioTrack::AudioTrack( 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); } -// DEPRECATED -AudioTrack::AudioTrack( - int streamType, - uint32_t sampleRate, - int format, - int channelMask, - int frameCount, - uint32_t flags, - callback_t cbf, - void* user, - int notificationFrames, - int sessionId) - : mStatus(NO_INIT), - mIsTimed(false), - mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) -{ - mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, - (audio_channel_mask_t) channelMask, - frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames, - 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); -} - AudioTrack::AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, @@ -188,7 +166,7 @@ status_t AudioTrack::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + int frameCountInt, audio_output_flags_t flags, callback_t cbf, void* user, @@ -197,10 +175,17 @@ status_t AudioTrack::set( bool threadCanCallJava, int sessionId) { + // FIXME "int" here is legacy and will be replaced by size_t later + if (frameCountInt < 0) { + ALOGE("Invalid frame count %d", frameCountInt); + return BAD_VALUE; + } + size_t frameCount = frameCountInt; - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); - ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); + ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); AutoMutex lock(mLock); if (mAudioTrack != 0) { @@ -214,7 +199,7 @@ status_t AudioTrack::set( } if (sampleRate == 0) { - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { return NO_INIT; } @@ -307,6 +292,15 @@ status_t AudioTrack::set( mFormat = format; mChannelMask = channelMask; mChannelCount = channelCount; + + if (audio_is_linear_pcm(format)) { + mFrameSize = channelCount * audio_bytes_per_sample(format); + mFrameSizeAF = channelCount * sizeof(int16_t); + } else { + mFrameSize = sizeof(uint8_t); + mFrameSizeAF = sizeof(uint8_t); + } + mSharedBuffer = sharedBuffer; mMuted = false; mActive = false; @@ -318,7 +312,6 @@ status_t AudioTrack::set( mUpdatePeriod = 0; mFlushed = false; AudioSystem::acquireAudioSessionId(mSessionId); - mRestoreStatus = NO_ERROR; return NO_ERROR; } @@ -349,20 +342,11 @@ int AudioTrack::channelCount() const return mChannelCount; } -uint32_t AudioTrack::frameCount() const +size_t AudioTrack::frameCount() const { return mCblk->frameCount; } -size_t AudioTrack::frameSize() const -{ - if (audio_is_linear_pcm(mFormat)) { - return channelCount()*audio_bytes_per_sample(mFormat); - } else { - return sizeof(uint8_t); - } -} - sp<IMemory>& AudioTrack::sharedBuffer() { return mSharedBuffer; @@ -390,7 +374,7 @@ void AudioTrack::start() cblk->lock.lock(); cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; cblk->waitTimeMs = 0; - android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); + android_atomic_and(~CBLK_DISABLED, &cblk->flags); if (t != 0) { t->resume(); } else { @@ -399,19 +383,21 @@ void AudioTrack::start() androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } - ALOGV("start %p before lock cblk %p", this, mCblk); + ALOGV("start %p before lock cblk %p", this, cblk); status_t status = NO_ERROR; - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { cblk->lock.unlock(); ALOGV("mAudioTrack->start()"); status = mAudioTrack->start(); cblk->lock.lock(); if (status == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } - if (cblk->flags & CBLK_INVALID_MSK) { - status = restoreTrack_l(cblk, true); + if (cblk->flags & CBLK_INVALID) { + audio_track_cblk_t* temp = cblk; + status = restoreTrack_l(temp, true /*fromStart*/); + cblk = temp; } cblk->lock.unlock(); if (status != NO_ERROR) { @@ -528,14 +514,9 @@ status_t AudioTrack::setVolume(float left, float right) return NO_ERROR; } -void AudioTrack::getVolume(float* left, float* right) const +status_t AudioTrack::setVolume(float volume) { - if (left != NULL) { - *left = mVolume[LEFT]; - } - if (right != NULL) { - *right = mVolume[RIGHT]; - } + return setVolume(volume, volume); } status_t AudioTrack::setAuxEffectSendLevel(float level) @@ -560,9 +541,9 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const } } -status_t AudioTrack::setSampleRate(int rate) +status_t AudioTrack::setSampleRate(uint32_t rate) { - int afSamplingRate; + uint32_t afSamplingRate; if (mIsTimed) { return INVALID_OPERATION; @@ -572,7 +553,7 @@ status_t AudioTrack::setSampleRate(int rate) return NO_INIT; } // Resampler implementation limits input sampling rate to 2 x output sampling rate. - if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; + if (rate == 0 || rate > afSamplingRate*2 ) return BAD_VALUE; AutoMutex lock(mLock); mCblk->sampleRate = rate; @@ -582,7 +563,7 @@ status_t AudioTrack::setSampleRate(int rate) uint32_t AudioTrack::getSampleRate() const { if (mIsTimed) { - return INVALID_OPERATION; + return 0; } AutoMutex lock(mLock); @@ -617,12 +598,14 @@ status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCou if (loopStart >= loopEnd || loopEnd - loopStart > cblk->frameCount || cblk->server > loopStart) { - ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); + ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " + "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); return BAD_VALUE; } if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { - ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", + ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " + "framecount %d", loopStart, loopEnd, cblk->frameCount); return BAD_VALUE; } @@ -683,12 +666,13 @@ status_t AudioTrack::setPosition(uint32_t position) if (!stopped_l()) return INVALID_OPERATION; - Mutex::Autolock _l(mCblk->lock); + audio_track_cblk_t* cblk = mCblk; + Mutex::Autolock _l(cblk->lock); - if (position > mCblk->user) return BAD_VALUE; + if (position > cblk->user) return BAD_VALUE; - mCblk->server = position; - android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); + cblk->server = position; + android_atomic_or(CBLK_FORCEREADY, &cblk->flags); return NO_ERROR; } @@ -710,7 +694,8 @@ status_t AudioTrack::reload() flush_l(); - mCblk->stepUser(mCblk->frameCount); + audio_track_cblk_t* cblk = mCblk; + cblk->stepUserOut(cblk->frameCount); return NO_ERROR; } @@ -751,7 +736,7 @@ status_t AudioTrack::createTrack_l( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output) @@ -791,7 +776,7 @@ status_t AudioTrack::createTrack_l( // Same comment as below about ignoring frameCount parameter for set() frameCount = sharedBuffer->size(); } else if (frameCount == 0) { - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { return NO_INIT; } @@ -823,11 +808,11 @@ status_t AudioTrack::createTrack_l( } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { // FIXME move these calculations and associated checks to server - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { return NO_INIT; } - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { return NO_INIT; } @@ -836,8 +821,8 @@ status_t AudioTrack::createTrack_l( uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); if (minBufCount < 2) minBufCount = 2; - int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; - ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" + size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; + ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" ", afLatency=%d", minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); @@ -849,7 +834,7 @@ status_t AudioTrack::createTrack_l( } // Make sure that application is notified with sufficient margin // before underrun - if (mNotificationFramesAct > (uint32_t)frameCount/2) { + if (mNotificationFramesAct > frameCount/2) { mNotificationFramesAct = frameCount/2; } if (frameCount < minFrameCount) { @@ -879,10 +864,12 @@ status_t AudioTrack::createTrack_l( sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), streamType, sampleRate, - format, + // AudioFlinger only sees 16-bit PCM + format == AUDIO_FORMAT_PCM_8_BIT ? + AUDIO_FORMAT_PCM_16_BIT : format, channelMask, frameCount, - trackFlags, + &trackFlags, sharedBuffer, output, tid, @@ -893,49 +880,49 @@ status_t AudioTrack::createTrack_l( ALOGE("AudioFlinger could not create track, status: %d", status); return status; } - sp<IMemory> cblk = track->getCblk(); - if (cblk == 0) { + sp<IMemory> iMem = track->getCblk(); + if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } mAudioTrack = track; - mCblkMemory = cblk; - mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); - // old has the previous value of mCblk->flags before the "or" operation - int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); + mCblkMemory = iMem; + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); + mCblk = cblk; if (flags & AUDIO_OUTPUT_FLAG_FAST) { - if (old & CBLK_FAST) { - ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); + if (trackFlags & IAudioFlinger::TRACK_FAST) { + ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", cblk->frameCount); } else { - ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); + ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", cblk->frameCount); // once denied, do not request again if IAudioTrack is re-created flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); mFlags = flags; } if (sharedBuffer == 0) { - mNotificationFramesAct = mCblk->frameCount/2; + mNotificationFramesAct = cblk->frameCount/2; } } if (sharedBuffer == 0) { - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); } else { - mCblk->buffers = sharedBuffer->pointer(); + mBuffers = sharedBuffer->pointer(); // Force buffer full condition as data is already present in shared memory - mCblk->stepUser(mCblk->frameCount); + cblk->stepUserOut(cblk->frameCount); } - mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); - mCblk->setSendLevel(mSendLevel); + cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | + uint16_t(mVolume[LEFT] * 0x1000)); + cblk->setSendLevel(mSendLevel); mAudioTrack->attachAuxEffect(mAuxEffectId); - mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - mCblk->waitTimeMs = 0; + cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; + cblk->waitTimeMs = 0; mRemainingFrames = mNotificationFramesAct; // FIXME don't believe this lie - mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; + mLatency = afLatency + (1000*cblk->frameCount) / sampleRate; // If IAudioTrack is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). - if (mCblk->frameCount > mFrameCount) { - mFrameCount = mCblk->frameCount; + if (cblk->frameCount > mFrameCount) { + mFrameCount = cblk->frameCount; } return NO_ERROR; } @@ -952,10 +939,10 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) audioBuffer->frameCount = 0; audioBuffer->size = 0; - uint32_t framesAvail = cblk->framesAvailable(); + uint32_t framesAvail = cblk->framesAvailableOut(); cblk->lock.lock(); - if (cblk->flags & CBLK_INVALID_MSK) { + if (cblk->flags & CBLK_INVALID) { goto create_new_track; } cblk->lock.unlock(); @@ -974,18 +961,23 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) cblk->lock.unlock(); return WOULD_BLOCK; } - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { mLock.unlock(); + // this condition is in shared memory, so if IAudioTrack and control block + // are replaced due to mediaserver death or IAudioTrack invalidation then + // cv won't be signalled, but fortunately the timeout will limit the wait result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); cblk->lock.unlock(); mLock.lock(); if (!mActive) { return status_t(STOPPED); } + // IAudioTrack may have been re-created while mLock was unlocked + cblk = mCblk; cblk->lock.lock(); } - if (cblk->flags & CBLK_INVALID_MSK) { + if (cblk->flags & CBLK_INVALID) { goto create_new_track; } if (CC_UNLIKELY(result != NO_ERROR)) { @@ -994,16 +986,18 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) // timing out when a loop has been set and we have already written upto loop end // is a normal condition: no need to wake AudioFlinger up. if (cblk->user < cblk->loopEnd) { - ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x" - "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server); + ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " + "server=%08x", this, cblk->mName, cblk->user, cblk->server); //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) cblk->lock.unlock(); result = mAudioTrack->start(); cblk->lock.lock(); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); create_new_track: - result = restoreTrack_l(cblk, false); + audio_track_cblk_t* temp = cblk; + result = restoreTrack_l(temp, false /*fromStart*/); + cblk = temp; } if (result != NO_ERROR) { ALOGW("obtainBuffer create Track error %d", result); @@ -1021,7 +1015,7 @@ create_new_track: } // read the server count again start_loop_here: - framesAvail = cblk->framesAvailable_l(); + framesAvail = cblk->framesAvailableOut_l(); } cblk->lock.unlock(); } @@ -1039,16 +1033,9 @@ create_new_track: framesReq = bufferEnd - u; } - audioBuffer->flags = mMuted ? Buffer::MUTE : 0; - audioBuffer->channelCount = mChannelCount; audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * cblk->frameSize; - if (audio_is_linear_pcm(mFormat)) { - audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; - } else { - audioBuffer->format = mFormat; - } - audioBuffer->raw = (int8_t *)cblk->buffer(u); + audioBuffer->size = framesReq * mFrameSizeAF; + audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } @@ -1056,12 +1043,13 @@ create_new_track: void AudioTrack::releaseBuffer(Buffer* audioBuffer) { AutoMutex lock(mLock); - mCblk->stepUser(audioBuffer->frameCount); + audio_track_cblk_t* cblk = mCblk; + cblk->stepUserOut(audioBuffer->frameCount); if (audioBuffer->frameCount > 0) { // restart track if it was disabled by audioflinger due to previous underrun - if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { - android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); - ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); + if (mActive && (cblk->flags & CBLK_DISABLED)) { + android_atomic_and(~CBLK_DISABLED, &cblk->flags); + ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); mAudioTrack->start(); } } @@ -1096,6 +1084,9 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) sp<IMemory> iMem = mCblkMemory; mLock.unlock(); + // since mLock is unlocked the IAudioTrack and shared memory may be re-created, + // so all cblk references might still refer to old shared memory, but that should be benign + ssize_t written = 0; const int8_t *src = (const int8_t *)buffer; Buffer audioBuffer; @@ -1121,8 +1112,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) } else { toWrite = audioBuffer.size; memcpy(audioBuffer.i8, src, toWrite); - src += toWrite; } + src += toWrite; userSize -= toWrite; written += toWrite; @@ -1140,24 +1131,33 @@ TimedAudioTrack::TimedAudioTrack() { status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) { + AutoMutex lock(mLock); status_t result = UNKNOWN_ERROR; + // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed + // while we are accessing the cblk + sp<IAudioTrack> audioTrack = mAudioTrack; + sp<IMemory> iMem = mCblkMemory; + // If the track is not invalid already, try to allocate a buffer. alloc // fails indicating that the server is dead, flag the track as invalid so // we can attempt to restore in just a bit. - if (!(mCblk->flags & CBLK_INVALID_MSK)) { + audio_track_cblk_t* cblk = mCblk; + if (!(cblk->flags & CBLK_INVALID)) { result = mAudioTrack->allocateTimedBuffer(size, buffer); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } // If the track is invalid at this point, attempt to restore it. and try the // allocation one more time. - if (mCblk->flags & CBLK_INVALID_MSK) { - mCblk->lock.lock(); - result = restoreTrack_l(mCblk, false); - mCblk->lock.unlock(); + if (cblk->flags & CBLK_INVALID) { + cblk->lock.lock(); + audio_track_cblk_t* temp = cblk; + result = restoreTrack_l(temp, false /*fromStart*/); + cblk = temp; + cblk->lock.unlock(); if (result == OK) result = mAudioTrack->allocateTimedBuffer(size, buffer); @@ -1172,10 +1172,11 @@ status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); { AutoMutex lock(mLock); + audio_track_cblk_t* cblk = mCblk; // restart track if it was disabled by audioflinger due to previous underrun if (buffer->size() != 0 && status == NO_ERROR && - mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { - android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); + mActive && (cblk->flags & CBLK_DISABLED)) { + android_atomic_and(~CBLK_DISABLED, &cblk->flags); ALOGW("queueTimedBuffer() track %p disabled, restarting", this); mAudioTrack->start(); } @@ -1206,10 +1207,13 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) bool active = mActive; mLock.unlock(); + // since mLock is unlocked the IAudioTrack and shared memory may be re-created, + // so all cblk references might still refer to old shared memory, but that should be benign + // Manage underrun callback - if (active && (cblk->framesAvailable() == cblk->frameCount)) { + if (active && (cblk->framesAvailableOut() == cblk->frameCount)) { ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { + if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { mCbf(EVENT_UNDERRUN, mUserData, 0); if (cblk->server == cblk->frameCount) { mCbf(EVENT_BUFFER_END, mUserData, 0); @@ -1265,7 +1269,8 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) status_t err = obtainBuffer(&audioBuffer, waitCount); if (err < NO_ERROR) { if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); + ALOGE_IF(err != status_t(NO_MORE_BUFFERS), + "Error obtaining an audio buffer, giving up."); return false; } break; @@ -1302,10 +1307,10 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) } audioBuffer.size = writtenSize; - // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for - // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of + // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for + // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of // 16 bit. - audioBuffer.frameCount = writtenSize/mCblk->frameSize; + audioBuffer.frameCount = writtenSize / mFrameSizeAF; frames -= audioBuffer.frameCount; @@ -1321,110 +1326,90 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) return true; } -// must be called with mLock and cblk.lock held. Callers must also hold strong references on +// must be called with mLock and refCblk.lock held. Callers must also hold strong references on // the IAudioTrack and IMemory in case they are recreated here. -// If the IAudioTrack is successfully restored, the cblk pointer is updated -status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) +// If the IAudioTrack is successfully restored, the refCblk pointer is updated +// FIXME Don't depend on caller to hold strong references. +status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) { status_t result; - if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { - ALOGW("dead IAudioTrack, creating a new one from %s TID %d", - fromStart ? "start()" : "obtainBuffer()", gettid()); + audio_track_cblk_t* cblk = refCblk; + audio_track_cblk_t* newCblk = cblk; + ALOGW("dead IAudioTrack, creating a new one from %s TID %d", + fromStart ? "start()" : "obtainBuffer()", gettid()); - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); + // signal old cblk condition so that other threads waiting for available buffers stop + // waiting now + cblk->cv.broadcast(); + cblk->lock.unlock(); - // refresh the audio configuration cache in this process to make sure we get new - // output parameters in getOutput_l() and createTrack_l() - AudioSystem::clearAudioConfigCache(); - - // if the new IAudioTrack is created, createTrack_l() will modify the - // following member variables: mAudioTrack, mCblkMemory and mCblk. - // It will also delete the strong references on previous IAudioTrack and IMemory - result = createTrack_l(mStreamType, - cblk->sampleRate, - mFormat, - mChannelMask, - mFrameCount, - mFlags, - mSharedBuffer, - getOutput_l()); - - if (result == NO_ERROR) { - uint32_t user = cblk->user; - uint32_t server = cblk->server; - // restore write index and set other indexes to reflect empty buffer status - mCblk->user = user; - mCblk->server = user; - mCblk->userBase = user; - mCblk->serverBase = user; - // restore loop: this is not guaranteed to succeed if new frame count is not - // compatible with loop length - setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); - if (!fromStart) { - mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - // Make sure that a client relying on callback events indicating underrun or - // the actual amount of audio frames played (e.g SoundPool) receives them. - if (mSharedBuffer == 0) { - uint32_t frames = 0; - if (user > server) { - frames = ((user - server) > mCblk->frameCount) ? - mCblk->frameCount : (user - server); - memset(mCblk->buffers, 0, frames * mCblk->frameSize); - } - // restart playback even if buffer is not completely filled. - android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); - // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to - // the client - mCblk->stepUser(frames); + // refresh the audio configuration cache in this process to make sure we get new + // output parameters in getOutput_l() and createTrack_l() + AudioSystem::clearAudioConfigCache(); + + // if the new IAudioTrack is created, createTrack_l() will modify the + // following member variables: mAudioTrack, mCblkMemory and mCblk. + // It will also delete the strong references on previous IAudioTrack and IMemory + result = createTrack_l(mStreamType, + cblk->sampleRate, + mFormat, + mChannelMask, + mFrameCount, + mFlags, + mSharedBuffer, + getOutput_l()); + + if (result == NO_ERROR) { + uint32_t user = cblk->user; + uint32_t server = cblk->server; + // restore write index and set other indexes to reflect empty buffer status + newCblk = mCblk; + newCblk->user = user; + newCblk->server = user; + newCblk->userBase = user; + newCblk->serverBase = user; + // restore loop: this is not guaranteed to succeed if new frame count is not + // compatible with loop length + setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); + if (!fromStart) { + newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + // Make sure that a client relying on callback events indicating underrun or + // the actual amount of audio frames played (e.g SoundPool) receives them. + if (mSharedBuffer == 0) { + uint32_t frames = 0; + if (user > server) { + frames = ((user - server) > newCblk->frameCount) ? + newCblk->frameCount : (user - server); + memset(mBuffers, 0, frames * mFrameSizeAF); } + // restart playback even if buffer is not completely filled. + android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); + // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to + // the client + newCblk->stepUserOut(frames); } - if (mSharedBuffer != 0) { - mCblk->stepUser(mCblk->frameCount); - } - if (mActive) { - result = mAudioTrack->start(); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); - } - if (fromStart && result == NO_ERROR) { - mNewPosition = mCblk->server + mUpdatePeriod; - } } - if (result != NO_ERROR) { - android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); + if (mSharedBuffer != 0) { + newCblk->stepUserOut(newCblk->frameCount); + } + if (mActive) { + result = mAudioTrack->start(); + ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); } - mRestoreStatus = result; - // signal old cblk condition for other threads waiting for restore completion - android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); - cblk->cv.broadcast(); - } else { - if (!(cblk->flags & CBLK_RESTORED_MSK)) { - ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); - mLock.unlock(); - result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); - if (result == NO_ERROR) { - result = mRestoreStatus; - } - cblk->lock.unlock(); - mLock.lock(); - } else { - ALOGW("dead IAudioTrack, already restored TID %d", gettid()); - result = mRestoreStatus; - cblk->lock.unlock(); + if (fromStart && result == NO_ERROR) { + mNewPosition = newCblk->server + mUpdatePeriod; } } + ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); + result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); if (result == NO_ERROR) { // from now on we switch to the newly created cblk - cblk = mCblk; + refCblk = newCblk; } - cblk->lock.lock(); + newCblk->lock.lock(); ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); @@ -1438,12 +1423,16 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const char buffer[SIZE]; String8 result; + audio_track_cblk_t* cblk = mCblk; result.append(" AudioTrack::dump\n"); - snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); + snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, + mVolume[0], mVolume[1]); result.append(buffer); - snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount); + snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, + mChannelCount, cblk->frameCount); result.append(buffer); - snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); + snprintf(buffer, 255, " sample rate(%u), status(%d), muted(%d)\n", + (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted); result.append(buffer); snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); result.append(buffer); @@ -1505,20 +1494,20 @@ void AudioTrack::AudioTrackThread::resume() audio_track_cblk_t::audio_track_cblk_t() : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), - userBase(0), serverBase(0), buffers(NULL), frameCount(0), + userBase(0), serverBase(0), frameCount(0), loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), mSendLevel(0), flags(0) { } -uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) +uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount, bool isOut) { ALOGV("stepuser %08x %08x %d", user, server, frameCount); uint32_t u = user; u += frameCount; // Ensure that user is never ahead of server for AudioRecord - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { // If stepServer() has been called once, switch to normal obtainBuffer() timeout period if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; @@ -1542,14 +1531,14 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) user = u; // Clear flow control error condition as new data has been written/read to/from buffer. - if (flags & CBLK_UNDERRUN_MSK) { - android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); + if (flags & CBLK_UNDERRUN) { + android_atomic_and(~CBLK_UNDERRUN, &flags); } return u; } -bool audio_track_cblk_t::stepServer(uint32_t frameCount) +bool audio_track_cblk_t::stepServer(uint32_t frameCount, bool isOut) { ALOGV("stepserver %08x %08x %d", user, server, frameCount); @@ -1562,7 +1551,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount) bool flushed = (s == user); s += frameCount; - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { // Mark that we have read the first buffer so that next time stepUser() is called // we switch to normal obtainBuffer() timeout period if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { @@ -1600,30 +1589,30 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount) server = s; - if (!(flags & CBLK_INVALID_MSK)) { + if (!(flags & CBLK_INVALID)) { cv.signal(); } lock.unlock(); return true; } -void* audio_track_cblk_t::buffer(uint32_t offset) const +void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const { return (int8_t *)buffers + (offset - userBase) * frameSize; } -uint32_t audio_track_cblk_t::framesAvailable() +uint32_t audio_track_cblk_t::framesAvailable(bool isOut) { Mutex::Autolock _l(lock); - return framesAvailable_l(); + return framesAvailable_l(isOut); } -uint32_t audio_track_cblk_t::framesAvailable_l() +uint32_t audio_track_cblk_t::framesAvailable_l(bool isOut) { uint32_t u = user; uint32_t s = server; - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { uint32_t limit = (s < loopStart) ? s : loopStart; return limit + frameCount - u; } else { @@ -1631,12 +1620,12 @@ uint32_t audio_track_cblk_t::framesAvailable_l() } } -uint32_t audio_track_cblk_t::framesReady() +uint32_t audio_track_cblk_t::framesReady(bool isOut) { uint32_t u = user; uint32_t s = server; - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { if (u < loopEnd) { return u - s; } else { diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index ce8ffc4..79c3361 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -89,8 +89,8 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, - track_flags_t flags, + size_t frameCount, + track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, @@ -106,7 +106,8 @@ public: data.writeInt32(format); data.writeInt32(channelMask); data.writeInt32(frameCount); - data.writeInt32((int32_t) flags); + track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT; + data.writeInt32(lFlags); data.writeStrongBinder(sharedBuffer->asBinder()); data.writeInt32((int32_t) output); data.writeInt32((int32_t) tid); @@ -119,6 +120,10 @@ public: if (lStatus != NO_ERROR) { ALOGE("createTrack error: %s", strerror(-lStatus)); } else { + lFlags = reply.readInt32(); + if (flags != NULL) { + *flags = lFlags; + } lSessionId = reply.readInt32(); if (sessionId != NULL) { *sessionId = lSessionId; @@ -138,7 +143,7 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, track_flags_t flags, pid_t tid, int *sessionId, @@ -522,7 +527,7 @@ public: return status; } - virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const + virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -690,7 +695,7 @@ public: return (audio_module_handle_t) reply.readInt32(); } - virtual int32_t getPrimaryOutputSamplingRate() + virtual uint32_t getPrimaryOutputSamplingRate() { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -698,7 +703,7 @@ public: return reply.readInt32(); } - virtual int32_t getPrimaryOutputFrameCount() + virtual size_t getPrimaryOutputFrameCount() { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -723,7 +728,7 @@ status_t BnAudioFlinger::onTransact( uint32_t sampleRate = data.readInt32(); audio_format_t format = (audio_format_t) data.readInt32(); audio_channel_mask_t channelMask = data.readInt32(); - size_t bufferCount = data.readInt32(); + size_t frameCount = data.readInt32(); track_flags_t flags = (track_flags_t) data.readInt32(); sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder()); audio_io_handle_t output = (audio_io_handle_t) data.readInt32(); @@ -732,7 +737,8 @@ status_t BnAudioFlinger::onTransact( status_t status; sp<IAudioTrack> track = createTrack(pid, (audio_stream_type_t) streamType, sampleRate, format, - channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status); + channelMask, frameCount, &flags, buffer, output, tid, &sessionId, &status); + reply->writeInt32(flags); reply->writeInt32(sessionId); reply->writeInt32(status); reply->writeStrongBinder(track->asBinder()); @@ -745,13 +751,13 @@ status_t BnAudioFlinger::onTransact( uint32_t sampleRate = data.readInt32(); audio_format_t format = (audio_format_t) data.readInt32(); audio_channel_mask_t channelMask = data.readInt32(); - size_t bufferCount = data.readInt32(); + size_t frameCount = data.readInt32(); track_flags_t flags = (track_flags_t) data.readInt32(); pid_t tid = (pid_t) data.readInt32(); int sessionId = data.readInt32(); status_t status; sp<IAudioRecord> record = openRecord(pid, input, - sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status); + sampleRate, format, channelMask, frameCount, flags, tid, &sessionId, &status); reply->writeInt32(sessionId); reply->writeInt32(status); reply->writeStrongBinder(record->asBinder()); @@ -865,7 +871,8 @@ status_t BnAudioFlinger::onTransact( case REGISTER_CLIENT: { CHECK_INTERFACE(IAudioFlinger, data, reply); - sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(data.readStrongBinder()); + sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>( + data.readStrongBinder()); registerClient(client); return NO_ERROR; } break; @@ -965,8 +972,8 @@ status_t BnAudioFlinger::onTransact( case GET_RENDER_POSITION: { CHECK_INTERFACE(IAudioFlinger, data, reply); audio_io_handle_t output = (audio_io_handle_t) data.readInt32(); - uint32_t halFrames; - uint32_t dspFrames; + size_t halFrames; + size_t dspFrames; status_t status = getRenderPosition(&halFrames, &dspFrames, output); reply->writeInt32(status); if (status == NO_ERROR) { @@ -1043,7 +1050,8 @@ status_t BnAudioFlinger::onTransact( int id; int enabled; - sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, &status, &id, &enabled); + sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, + &status, &id, &enabled); reply->writeInt32(status); reply->writeInt32(id); reply->writeInt32(enabled); diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp index 4178b29..2d1e0f8 100644 --- a/media/libmedia/IAudioFlingerClient.cpp +++ b/media/libmedia/IAudioFlingerClient.cpp @@ -50,7 +50,8 @@ public: ALOGV("ioConfigChanged stream %d", stream); data.writeInt32(stream); } else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) { - const AudioSystem::OutputDescriptor *desc = (const AudioSystem::OutputDescriptor *)param2; + const AudioSystem::OutputDescriptor *desc = + (const AudioSystem::OutputDescriptor *)param2; data.writeInt32(desc->samplingRate); data.writeInt32(desc->format); data.writeInt32(desc->channels); diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index 401437c..769deae 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -399,13 +399,15 @@ status_t BnAudioPolicyService::onTransact( case SET_PHONE_STATE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32()))); + reply->writeInt32(static_cast <uint32_t>(setPhoneState( + (audio_mode_t) data.readInt32()))); return NO_ERROR; } break; case SET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32()); + audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>( + data.readInt32()); audio_policy_forced_cfg_t config = static_cast <audio_policy_forced_cfg_t>(data.readInt32()); reply->writeInt32(static_cast <uint32_t>(setForceUse(usage, config))); @@ -414,7 +416,8 @@ status_t BnAudioPolicyService::onTransact( case GET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32()); + audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>( + data.readInt32()); reply->writeInt32(static_cast <uint32_t>(getForceUse(usage))); return NO_ERROR; } break; diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp index abc8899..204e0ce 100644 --- a/media/libmedia/SoundPool.cpp +++ b/media/libmedia/SoundPool.cpp @@ -568,8 +568,8 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV } // initialize track - int afFrameCount; - int afSampleRate; + size_t afFrameCount; + uint32_t afSampleRate; audio_stream_type_t streamType = mSoundPool->streamType(); if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { afFrameCount = kDefaultFrameCount; diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp index 253602d..42584fe 100644 --- a/media/libmedia/ToneGenerator.cpp +++ b/media/libmedia/ToneGenerator.cpp @@ -1036,7 +1036,7 @@ bool ToneGenerator::initAudioTrack() { goto initAudioTrack_exit; } - mpAudioTrack->setVolume(mVolume, mVolume); + mpAudioTrack->setVolume(mVolume); mState = TONE_INIT; diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp index 8196e10..5b4071b 100644 --- a/media/libmedia/Visualizer.cpp +++ b/media/libmedia/Visualizer.cpp @@ -88,7 +88,8 @@ status_t Visualizer::setEnabled(bool enabled) return status; } -status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, uint32_t rate) +status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, + uint32_t rate) { if (rate > CAPTURE_RATE_MAX) { return BAD_VALUE; @@ -334,7 +335,8 @@ void Visualizer::controlStatusChanged(bool controlGranted) { //------------------------------------------------------------------------- -Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate, bool bCanCallJava) +Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate, + bool bCanCallJava) : Thread(bCanCallJava), mReceiver(receiver) { mSleepTimeUs = 1000000000 / captureRate; diff --git a/media/libmedia_native/Android.mk b/media/libmedia_native/Android.mk deleted file mode 100644 index 065a90f..0000000 --- a/media/libmedia_native/Android.mk +++ /dev/null @@ -1,11 +0,0 @@ -LOCAL_PATH := $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES := - -LOCAL_MODULE:= libmedia_native - -LOCAL_MODULE_TAGS := optional - -include $(BUILD_SHARED_LIBRARY) diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk index 5b5ed71..48f48e4 100644 --- a/media/libmediaplayerservice/Android.mk +++ b/media/libmediaplayerservice/Android.mk @@ -28,7 +28,6 @@ LOCAL_SHARED_LIBRARIES := \ libdl \ libgui \ libmedia \ - libmedia_native \ libsonivox \ libstagefright \ libstagefright_foundation \ diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp index 9bedff1..c3e5c40 100644 --- a/media/libmediaplayerservice/MediaPlayerService.cpp +++ b/media/libmediaplayerservice/MediaPlayerService.cpp @@ -1387,8 +1387,8 @@ status_t MediaPlayerService::AudioOutput::open( } ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount, mSessionId); - int afSampleRate; - int afFrameCount; + uint32_t afSampleRate; + size_t afFrameCount; uint32_t frameCount; if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) { diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp index 1ddf775..756e76a 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp @@ -513,8 +513,6 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { } } } else if (what == Renderer::kWhatFlushComplete) { - CHECK_EQ(what, (int32_t)Renderer::kWhatFlushComplete); - int32_t audio; CHECK(msg->findInt32("audio", &audio)); diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp index 00d2017..e0d2c21 100644 --- a/media/libnbaio/NBAIO.cpp +++ b/media/libnbaio/NBAIO.cpp @@ -24,44 +24,55 @@ namespace android { size_t Format_frameSize(NBAIO_Format format) { - switch (format) { - case Format_SR44_1_C2_I16: - case Format_SR48_C2_I16: - return 2 * sizeof(short); - case Format_SR44_1_C1_I16: - case Format_SR48_C1_I16: - return 1 * sizeof(short); - case Format_Invalid: - default: - return 0; - } + return Format_channelCount(format) * sizeof(short); } size_t Format_frameBitShift(NBAIO_Format format) { - switch (format) { - case Format_SR44_1_C2_I16: - case Format_SR48_C2_I16: - return 2; // 1 << 2 == 2 * sizeof(short) - case Format_SR44_1_C1_I16: - case Format_SR48_C1_I16: - return 1; // 1 << 1 == 1 * sizeof(short) - case Format_Invalid: - default: - return 0; - } + // sizeof(short) == 2, so frame size == 1 << channels + return Format_channelCount(format); } +enum { + Format_SR_8000, + Format_SR_11025, + Format_SR_16000, + Format_SR_22050, + Format_SR_24000, + Format_SR_32000, + Format_SR_44100, + Format_SR_48000, + Format_SR_Mask = 7 +}; + +enum { + Format_C_1 = 0x08, + Format_C_2 = 0x10, + Format_C_Mask = 0x18 +}; + unsigned Format_sampleRate(NBAIO_Format format) { - switch (format) { - case Format_SR44_1_C1_I16: - case Format_SR44_1_C2_I16: + if (format == Format_Invalid) { + return 0; + } + switch (format & Format_SR_Mask) { + case Format_SR_8000: + return 8000; + case Format_SR_11025: + return 11025; + case Format_SR_16000: + return 16000; + case Format_SR_22050: + return 22050; + case Format_SR_24000: + return 24000; + case Format_SR_32000: + return 32000; + case Format_SR_44100: return 44100; - case Format_SR48_C1_I16: - case Format_SR48_C2_I16: + case Format_SR_48000: return 48000; - case Format_Invalid: default: return 0; } @@ -69,14 +80,14 @@ unsigned Format_sampleRate(NBAIO_Format format) unsigned Format_channelCount(NBAIO_Format format) { - switch (format) { - case Format_SR44_1_C1_I16: - case Format_SR48_C1_I16: + if (format == Format_Invalid) { + return 0; + } + switch (format & Format_C_Mask) { + case Format_C_1: return 1; - case Format_SR44_1_C2_I16: - case Format_SR48_C2_I16: + case Format_C_2: return 2; - case Format_Invalid: default: return 0; } @@ -84,11 +95,46 @@ unsigned Format_channelCount(NBAIO_Format format) NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount) { - if (sampleRate == 44100 && channelCount == 2) return Format_SR44_1_C2_I16; - if (sampleRate == 48000 && channelCount == 2) return Format_SR48_C2_I16; - if (sampleRate == 44100 && channelCount == 1) return Format_SR44_1_C1_I16; - if (sampleRate == 48000 && channelCount == 1) return Format_SR48_C1_I16; - return Format_Invalid; + NBAIO_Format format; + switch (sampleRate) { + case 8000: + format = Format_SR_8000; + break; + case 11025: + format = Format_SR_11025; + break; + case 16000: + format = Format_SR_16000; + break; + case 22050: + format = Format_SR_22050; + break; + case 24000: + format = Format_SR_24000; + break; + case 32000: + format = Format_SR_32000; + break; + case 44100: + format = Format_SR_44100; + break; + case 48000: + format = Format_SR_48000; + break; + default: + return Format_Invalid; + } + switch (channelCount) { + case 1: + format |= Format_C_1; + break; + case 2: + format |= Format_C_2; + break; + default: + return Format_Invalid; + } + return format; } // This is a default implementation; it is expected that subclasses will optimize this. diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 84b4962..a135222 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -803,6 +803,8 @@ status_t ACodec::setComponentRole( "audio_decoder.raw", "audio_encoder.raw" }, { MEDIA_MIMETYPE_AUDIO_FLAC, "audio_decoder.flac", "audio_encoder.flac" }, + { MEDIA_MIMETYPE_AUDIO_MSGSM, + "audio_decoder.gsm", "audio_encoder.gsm" }, }; static const size_t kNumMimeToRole = diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index faa0f31..a056706 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -78,7 +78,6 @@ LOCAL_SHARED_LIBRARIES := \ libicuuc \ liblog \ libmedia \ - libmedia_native \ libsonivox \ libssl \ libstagefright_omx \ diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp index 861aebe..3cf4d5c 100644 --- a/media/libstagefright/AudioSource.cpp +++ b/media/libstagefright/AudioSource.cpp @@ -58,7 +58,7 @@ AudioSource::AudioSource( ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount); CHECK(channelCount == 1 || channelCount == 2); - int minFrameCount; + size_t minFrameCount; status_t status = AudioRecord::getMinFrameCount(&minFrameCount, sampleRate, AUDIO_FORMAT_PCM_16_BIT, diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp index efd7af7..efd7af7 100755..100644 --- a/media/libstagefright/CameraSource.cpp +++ b/media/libstagefright/CameraSource.cpp diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp index 8b52e15..8b52e15 100755..100644 --- a/media/libstagefright/MPEG4Writer.cpp +++ b/media/libstagefright/MPEG4Writer.cpp diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp index e7b5903..5d8029c 100644 --- a/media/libstagefright/MediaDefs.cpp +++ b/media/libstagefright/MediaDefs.cpp @@ -40,6 +40,7 @@ const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW = "audio/g711-mlaw"; const char *MEDIA_MIMETYPE_AUDIO_RAW = "audio/raw"; const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac"; const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts"; +const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm"; const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4"; const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav"; diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp index 70de174..22aefcc 100755..100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -1390,6 +1390,8 @@ void OMXCodec::setComponentRole( "audio_decoder.raw", "audio_encoder.raw" }, { MEDIA_MIMETYPE_AUDIO_FLAC, "audio_decoder.flac", "audio_encoder.flac" }, + { MEDIA_MIMETYPE_AUDIO_MSGSM, + "audio_decoder.gsm", "audio_encoder.gsm" }, }; static const size_t kNumMimeToRole = diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp index 773854f..773854f 100755..100644 --- a/media/libstagefright/SkipCutBuffer.cpp +++ b/media/libstagefright/SkipCutBuffer.cpp diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp index bccffd8..af8186c 100644 --- a/media/libstagefright/StagefrightMediaScanner.cpp +++ b/media/libstagefright/StagefrightMediaScanner.cpp @@ -42,7 +42,7 @@ static bool FileHasAcceptableExtension(const char *extension) { ".mpeg", ".ogg", ".mid", ".smf", ".imy", ".wma", ".aac", ".wav", ".amr", ".midi", ".xmf", ".rtttl", ".rtx", ".ota", ".mkv", ".mka", ".webm", ".ts", ".fl", ".flac", ".mxmf", - ".avi", ".mpeg", ".mpg", ".mpga" + ".avi", ".mpeg", ".mpg", ".awb", ".mpga" }; static const size_t kNumValidExtensions = sizeof(kValidExtensions) / sizeof(kValidExtensions[0]); diff --git a/media/libstagefright/ThrottledSource.cpp b/media/libstagefright/ThrottledSource.cpp index 348a9d3..7496752 100644 --- a/media/libstagefright/ThrottledSource.cpp +++ b/media/libstagefright/ThrottledSource.cpp @@ -31,10 +31,6 @@ ThrottledSource::ThrottledSource( CHECK(mBandwidthLimitBytesPerSecond > 0); } -status_t ThrottledSource::initCheck() const { - return mSource->initCheck(); -} - ssize_t ThrottledSource::readAt(off64_t offset, void *data, size_t size) { Mutex::Autolock autoLock(mLock); @@ -62,17 +58,9 @@ ssize_t ThrottledSource::readAt(off64_t offset, void *data, size_t size) { if (whenUs > nowUs) { usleep(whenUs - nowUs); } - return n; } -status_t ThrottledSource::getSize(off64_t *size) { - return mSource->getSize(size); -} - -uint32_t ThrottledSource::flags() { - return mSource->flags(); -} } // namespace android diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp index a38400b..d32f4fb 100644 --- a/media/libstagefright/WAVExtractor.cpp +++ b/media/libstagefright/WAVExtractor.cpp @@ -38,6 +38,7 @@ enum { WAVE_FORMAT_PCM = 0x0001, WAVE_FORMAT_ALAW = 0x0006, WAVE_FORMAT_MULAW = 0x0007, + WAVE_FORMAT_MSGSM = 0x0031, WAVE_FORMAT_EXTENSIBLE = 0xFFFE }; @@ -178,6 +179,7 @@ status_t WAVExtractor::init() { if (mWaveFormat != WAVE_FORMAT_PCM && mWaveFormat != WAVE_FORMAT_ALAW && mWaveFormat != WAVE_FORMAT_MULAW + && mWaveFormat != WAVE_FORMAT_MSGSM && mWaveFormat != WAVE_FORMAT_EXTENSIBLE) { return ERROR_UNSUPPORTED; } @@ -216,6 +218,10 @@ status_t WAVExtractor::init() { && mBitsPerSample != 24) { return ERROR_UNSUPPORTED; } + } else if (mWaveFormat == WAVE_FORMAT_MSGSM) { + if (mBitsPerSample != 0) { + return ERROR_UNSUPPORTED; + } } else { CHECK(mWaveFormat == WAVE_FORMAT_MULAW || mWaveFormat == WAVE_FORMAT_ALAW); @@ -283,6 +289,10 @@ status_t WAVExtractor::init() { mTrackMeta->setCString( kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW); break; + case WAVE_FORMAT_MSGSM: + mTrackMeta->setCString( + kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MSGSM); + break; default: CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW); mTrackMeta->setCString( @@ -294,11 +304,17 @@ status_t WAVExtractor::init() { mTrackMeta->setInt32(kKeyChannelMask, mChannelMask); mTrackMeta->setInt32(kKeySampleRate, mSampleRate); - size_t bytesPerSample = mBitsPerSample >> 3; - - int64_t durationUs = - 1000000LL * (mDataSize / (mNumChannels * bytesPerSample)) - / mSampleRate; + int64_t durationUs = 0; + if (mWaveFormat == WAVE_FORMAT_MSGSM) { + // 65 bytes decode to 320 8kHz samples + durationUs = + 1000000LL * (mDataSize / 65 * 320) / 8000; + } else { + size_t bytesPerSample = mBitsPerSample >> 3; + durationUs = + 1000000LL * (mDataSize / (mNumChannels * bytesPerSample)) + / mSampleRate; + } mTrackMeta->setInt64(kKeyDuration, durationUs); @@ -388,7 +404,16 @@ status_t WAVSource::read( int64_t seekTimeUs; ReadOptions::SeekMode mode; if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) { - int64_t pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3); + int64_t pos = 0; + + if (mWaveFormat == WAVE_FORMAT_MSGSM) { + // 65 bytes decode to 320 8kHz samples + int64_t samplenumber = (seekTimeUs * mSampleRate) / 1000000; + int64_t framenumber = samplenumber / 320; + pos = framenumber * 65; + } else { + pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3); + } if (pos > mSize) { pos = mSize; } @@ -412,6 +437,15 @@ status_t WAVSource::read( maxBytesToRead = maxBytesAvailable; } + if (mWaveFormat == WAVE_FORMAT_MSGSM) { + // Microsoft packs 2 frames into 65 bytes, rather than using separate 33-byte frames, + // so read multiples of 65, and use smaller buffers to account for ~10:1 expansion ratio + if (maxBytesToRead > 1024) { + maxBytesToRead = 1024; + } + maxBytesToRead = (maxBytesToRead / 65) * 65; + } + ssize_t n = mDataSource->readAt( mCurrentPos, buffer->data(), maxBytesToRead); @@ -468,12 +502,17 @@ status_t WAVSource::read( } } - size_t bytesPerSample = mBitsPerSample >> 3; + int64_t timeStampUs = 0; + + if (mWaveFormat == WAVE_FORMAT_MSGSM) { + timeStampUs = 1000000LL * (mCurrentPos - mOffset) * 320 / 65 / mSampleRate; + } else { + size_t bytesPerSample = mBitsPerSample >> 3; + timeStampUs = 1000000LL * (mCurrentPos - mOffset) + / (mNumChannels * bytesPerSample) / mSampleRate; + } - buffer->meta_data()->setInt64( - kKeyTime, - 1000000LL * (mCurrentPos - mOffset) - / (mNumChannels * bytesPerSample) / mSampleRate); + buffer->meta_data()->setInt64(kKeyTime, timeStampUs); buffer->meta_data()->setInt32(kKeyIsSyncFrame, 1); mCurrentPos += n; diff --git a/media/libstagefright/codecs/gsm/Android.mk b/media/libstagefright/codecs/gsm/Android.mk new file mode 100644 index 0000000..2e43120 --- /dev/null +++ b/media/libstagefright/codecs/gsm/Android.mk @@ -0,0 +1,4 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/media/libstagefright/codecs/gsm/dec/Android.mk b/media/libstagefright/codecs/gsm/dec/Android.mk new file mode 100644 index 0000000..9c0c6ae --- /dev/null +++ b/media/libstagefright/codecs/gsm/dec/Android.mk @@ -0,0 +1,21 @@ +LOCAL_PATH:= $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_SRC_FILES := \ + SoftGSM.cpp + +LOCAL_C_INCLUDES := \ + frameworks/av/media/libstagefright/include \ + frameworks/native/include/media/openmax \ + external/libgsm/inc + +LOCAL_SHARED_LIBRARIES := \ + libstagefright libstagefright_omx libstagefright_foundation libutils + +LOCAL_STATIC_LIBRARIES := \ + libgsm + +LOCAL_MODULE := libstagefright_soft_gsmdec +LOCAL_MODULE_TAGS := optional + +include $(BUILD_SHARED_LIBRARY) diff --git a/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2 b/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2 new file mode 100644 index 0000000..e69de29 --- /dev/null +++ b/media/libstagefright/codecs/gsm/dec/MODULE_LICENSE_APACHE2 diff --git a/media/libstagefright/codecs/gsm/dec/NOTICE b/media/libstagefright/codecs/gsm/dec/NOTICE new file mode 100644 index 0000000..c5b1efa --- /dev/null +++ b/media/libstagefright/codecs/gsm/dec/NOTICE @@ -0,0 +1,190 @@ + + Copyright (c) 2005-2008, The Android Open Source Project + + Licensed under the Apache License, Version 2.0 (the "License"); + you may not use this file except in compliance with the License. + + Unless required by applicable law or agreed to in writing, software + distributed under the License is distributed on an "AS IS" BASIS, + WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + See the License for the specific language governing permissions and + limitations under the License. + + + Apache License + Version 2.0, January 2004 + http://www.apache.org/licenses/ + + TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION + + 1. 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However, in accepting such obligations, You may act only + on Your own behalf and on Your sole responsibility, not on behalf + of any other Contributor, and only if You agree to indemnify, + defend, and hold each Contributor harmless for any liability + incurred by, or claims asserted against, such Contributor by reason + of your accepting any such warranty or additional liability. + + END OF TERMS AND CONDITIONS + diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp new file mode 100644 index 0000000..00e0c85 --- /dev/null +++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.cpp @@ -0,0 +1,269 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "SoftGSM" +#include <utils/Log.h> + +#include "SoftGSM.h" + +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/MediaDefs.h> + +namespace android { + +template<class T> +static void InitOMXParams(T *params) { + params->nSize = sizeof(T); + params->nVersion.s.nVersionMajor = 1; + params->nVersion.s.nVersionMinor = 0; + params->nVersion.s.nRevision = 0; + params->nVersion.s.nStep = 0; +} + +SoftGSM::SoftGSM( + const char *name, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component) + : SimpleSoftOMXComponent(name, callbacks, appData, component), + mSignalledError(false) { + + CHECK(!strcmp(name, "OMX.google.gsm.decoder")); + + mGsm = gsm_create(); + CHECK(mGsm); + int msopt = 1; + gsm_option(mGsm, GSM_OPT_WAV49, &msopt); + + initPorts(); +} + +SoftGSM::~SoftGSM() { + gsm_destroy(mGsm); +} + +void SoftGSM::initPorts() { + OMX_PARAM_PORTDEFINITIONTYPE def; + InitOMXParams(&def); + + def.nPortIndex = 0; + def.eDir = OMX_DirInput; + def.nBufferCountMin = kNumBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.nBufferSize = sizeof(gsm_frame); + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainAudio; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 1; + + def.format.audio.cMIMEType = + const_cast<char *>(MEDIA_MIMETYPE_AUDIO_MSGSM); + + def.format.audio.pNativeRender = NULL; + def.format.audio.bFlagErrorConcealment = OMX_FALSE; + def.format.audio.eEncoding = OMX_AUDIO_CodingGSMFR; + + addPort(def); + + def.nPortIndex = 1; + def.eDir = OMX_DirOutput; + def.nBufferCountMin = kNumBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.nBufferSize = kMaxNumSamplesPerFrame * sizeof(int16_t); + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainAudio; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 2; + + def.format.audio.cMIMEType = const_cast<char *>("audio/raw"); + def.format.audio.pNativeRender = NULL; + def.format.audio.bFlagErrorConcealment = OMX_FALSE; + def.format.audio.eEncoding = OMX_AUDIO_CodingPCM; + + addPort(def); +} + +OMX_ERRORTYPE SoftGSM::internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params) { + switch (index) { + case OMX_IndexParamAudioPcm: + { + OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams = + (OMX_AUDIO_PARAM_PCMMODETYPE *)params; + + if (pcmParams->nPortIndex > 1) { + return OMX_ErrorUndefined; + } + + pcmParams->eNumData = OMX_NumericalDataSigned; + pcmParams->eEndian = OMX_EndianBig; + pcmParams->bInterleaved = OMX_TRUE; + pcmParams->nBitPerSample = 16; + pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear; + pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF; + pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF; + + pcmParams->nChannels = 1; + pcmParams->nSamplingRate = 8000; + + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalGetParameter(index, params); + } +} + +OMX_ERRORTYPE SoftGSM::internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params) { + switch (index) { + case OMX_IndexParamAudioPcm: + { + OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams = + (OMX_AUDIO_PARAM_PCMMODETYPE *)params; + + if (pcmParams->nPortIndex != 0 && pcmParams->nPortIndex != 1) { + return OMX_ErrorUndefined; + } + + if (pcmParams->nChannels != 1) { + return OMX_ErrorUndefined; + } + + if (pcmParams->nSamplingRate != 8000) { + return OMX_ErrorUndefined; + } + + return OMX_ErrorNone; + } + + case OMX_IndexParamStandardComponentRole: + { + const OMX_PARAM_COMPONENTROLETYPE *roleParams = + (const OMX_PARAM_COMPONENTROLETYPE *)params; + + if (strncmp((const char *)roleParams->cRole, + "audio_decoder.gsm", + OMX_MAX_STRINGNAME_SIZE - 1)) { + return OMX_ErrorUndefined; + } + + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalSetParameter(index, params); + } +} + +void SoftGSM::onQueueFilled(OMX_U32 portIndex) { + if (mSignalledError) { + return; + } + + List<BufferInfo *> &inQueue = getPortQueue(0); + List<BufferInfo *> &outQueue = getPortQueue(1); + + while (!inQueue.empty() && !outQueue.empty()) { + BufferInfo *inInfo = *inQueue.begin(); + OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + + BufferInfo *outInfo = *outQueue.begin(); + OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + inQueue.erase(inQueue.begin()); + inInfo->mOwnedByUs = false; + notifyEmptyBufferDone(inHeader); + + outHeader->nFilledLen = 0; + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + + outQueue.erase(outQueue.begin()); + outInfo->mOwnedByUs = false; + notifyFillBufferDone(outHeader); + return; + } + + if (inHeader->nFilledLen > kMaxNumSamplesPerFrame) { + ALOGE("input buffer too large (%ld).", inHeader->nFilledLen); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + mSignalledError = true; + } + + if(((inHeader->nFilledLen / 65) * 65) != inHeader->nFilledLen) { + ALOGE("input buffer not multiple of 65 (%ld).", inHeader->nFilledLen); + notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL); + mSignalledError = true; + } + + uint8_t *inputptr = inHeader->pBuffer + inHeader->nOffset; + + int n = mSignalledError ? 0 : DecodeGSM(mGsm, + reinterpret_cast<int16_t *>(outHeader->pBuffer), inputptr, inHeader->nFilledLen); + + outHeader->nTimeStamp = inHeader->nTimeStamp; + outHeader->nOffset = 0; + outHeader->nFilledLen = n * sizeof(int16_t); + outHeader->nFlags = 0; + + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + + outInfo->mOwnedByUs = false; + outQueue.erase(outQueue.begin()); + outInfo = NULL; + notifyFillBufferDone(outHeader); + outHeader = NULL; + } +} + + +// static +int SoftGSM::DecodeGSM(gsm handle, + int16_t *out, uint8_t *in, size_t inSize) { + + int ret = 0; + while (inSize > 0) { + gsm_decode(handle, in, out); + in += 33; + inSize -= 33; + out += 160; + ret += 160; + gsm_decode(handle, in, out); + in += 32; + inSize -= 32; + out += 160; + ret += 160; + } + return ret; +} + + +} // namespace android + +android::SoftOMXComponent *createSoftOMXComponent( + const char *name, const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, OMX_COMPONENTTYPE **component) { + return new android::SoftGSM(name, callbacks, appData, component); +} + diff --git a/media/libstagefright/codecs/gsm/dec/SoftGSM.h b/media/libstagefright/codecs/gsm/dec/SoftGSM.h new file mode 100644 index 0000000..8ab6116 --- /dev/null +++ b/media/libstagefright/codecs/gsm/dec/SoftGSM.h @@ -0,0 +1,65 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef SOFT_GSM_H_ + +#define SOFT_GSM_H_ + +#include "SimpleSoftOMXComponent.h" + +extern "C" { +#include "gsm.h" +} + +namespace android { + +struct SoftGSM : public SimpleSoftOMXComponent { + SoftGSM(const char *name, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component); + +protected: + virtual ~SoftGSM(); + + virtual OMX_ERRORTYPE internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params); + + virtual OMX_ERRORTYPE internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params); + + virtual void onQueueFilled(OMX_U32 portIndex); + +private: + enum { + kNumBuffers = 4, + kMaxNumSamplesPerFrame = 16384, + }; + + bool mSignalledError; + gsm mGsm; + + void initPorts(); + + static int DecodeGSM(gsm handle, int16_t *out, uint8_t *in, size_t inSize); + + DISALLOW_EVIL_CONSTRUCTORS(SoftGSM); +}; + +} // namespace android + +#endif // SOFT_GSM_H_ + diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp index bf9ab3a..a400b4c 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp @@ -66,7 +66,7 @@ void SoftVPX::initPorts() { def.eDir = OMX_DirInput; def.nBufferCountMin = kNumBuffers; def.nBufferCountActual = def.nBufferCountMin; - def.nBufferSize = 256 * 1024; + def.nBufferSize = 768 * 1024; def.bEnabled = OMX_TRUE; def.bPopulated = OMX_FALSE; def.eDomain = OMX_PortDomainVideo; diff --git a/media/libstagefright/include/FragmentedMP4Parser.h b/media/libstagefright/include/FragmentedMP4Parser.h index 0edafb9..dbe02b8 100644 --- a/media/libstagefright/include/FragmentedMP4Parser.h +++ b/media/libstagefright/include/FragmentedMP4Parser.h @@ -263,7 +263,7 @@ private: void copyBuffer( sp<ABuffer> *dst, - size_t offset, uint64_t size, size_t extra = 0) const; + size_t offset, uint64_t size) const; DISALLOW_EVIL_CONSTRUCTORS(FragmentedMP4Parser); }; diff --git a/media/libstagefright/include/ThrottledSource.h b/media/libstagefright/include/ThrottledSource.h index 7fe7c06..673268b 100644 --- a/media/libstagefright/include/ThrottledSource.h +++ b/media/libstagefright/include/ThrottledSource.h @@ -28,18 +28,44 @@ struct ThrottledSource : public DataSource { const sp<DataSource> &source, int32_t bandwidthLimitBytesPerSecond); - virtual status_t initCheck() const; - + // implementation of readAt() that sleeps to achieve the desired max throughput virtual ssize_t readAt(off64_t offset, void *data, size_t size); - virtual status_t getSize(off64_t *size); - virtual uint32_t flags(); + // returns an empty string to prevent callers from using the Uri to construct a new datasource + virtual String8 getUri() { + return String8(); + } + + // following methods all call through to the wrapped DataSource's methods + + status_t initCheck() const { + return mSource->initCheck(); + } + + virtual status_t getSize(off64_t *size) { + return mSource->getSize(size); + } + + virtual uint32_t flags() { + return mSource->flags(); + } + + virtual status_t reconnectAtOffset(off64_t offset) { + return mSource->reconnectAtOffset(offset); + } + + virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL) { + return mSource->DrmInitialization(mime); + } + + virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) { + mSource->getDrmInfo(handle, client); + }; virtual String8 getMIMEType() const { return mSource->getMIMEType(); } - private: Mutex mLock; diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp index 8f7d12b..7fc7037 100644 --- a/media/libstagefright/matroska/MatroskaExtractor.cpp +++ b/media/libstagefright/matroska/MatroskaExtractor.cpp @@ -758,31 +758,69 @@ static void addESDSFromCodecPrivate( esds = NULL; } -void addVorbisCodecInfo( +status_t addVorbisCodecInfo( const sp<MetaData> &meta, const void *_codecPrivate, size_t codecPrivateSize) { - // printf("vorbis private data follows:\n"); // hexdump(_codecPrivate, codecPrivateSize); - CHECK(codecPrivateSize >= 3); + if (codecPrivateSize < 1) { + return ERROR_MALFORMED; + } const uint8_t *codecPrivate = (const uint8_t *)_codecPrivate; - CHECK(codecPrivate[0] == 0x02); - size_t len1 = codecPrivate[1]; - size_t len2 = codecPrivate[2]; + if (codecPrivate[0] != 0x02) { + return ERROR_MALFORMED; + } - CHECK(codecPrivateSize > 3 + len1 + len2); + // codecInfo starts with two lengths, len1 and len2, that are + // "Xiph-style-lacing encoded"... - CHECK(codecPrivate[3] == 0x01); - meta->setData(kKeyVorbisInfo, 0, &codecPrivate[3], len1); + size_t offset = 1; + size_t len1 = 0; + while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) { + len1 += 0xff; + ++offset; + } + if (offset >= codecPrivateSize) { + return ERROR_MALFORMED; + } + len1 += codecPrivate[offset++]; - CHECK(codecPrivate[len1 + 3] == 0x03); + size_t len2 = 0; + while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) { + len2 += 0xff; + ++offset; + } + if (offset >= codecPrivateSize) { + return ERROR_MALFORMED; + } + len2 += codecPrivate[offset++]; + + if (codecPrivateSize < offset + len1 + len2) { + return ERROR_MALFORMED; + } + + if (codecPrivate[offset] != 0x01) { + return ERROR_MALFORMED; + } + meta->setData(kKeyVorbisInfo, 0, &codecPrivate[offset], len1); + + offset += len1; + if (codecPrivate[offset] != 0x03) { + return ERROR_MALFORMED; + } + + offset += len2; + if (codecPrivate[offset] != 0x05) { + return ERROR_MALFORMED; + } - CHECK(codecPrivate[len1 + len2 + 3] == 0x05); meta->setData( - kKeyVorbisBooks, 0, &codecPrivate[len1 + len2 + 3], - codecPrivateSize - len1 - len2 - 3); + kKeyVorbisBooks, 0, &codecPrivate[offset], + codecPrivateSize - offset); + + return OK; } void MatroskaExtractor::addTracks() { @@ -809,6 +847,8 @@ void MatroskaExtractor::addTracks() { sp<MetaData> meta = new MetaData; + status_t err = OK; + switch (track->GetType()) { case VIDEO_TRACK: { @@ -855,7 +895,8 @@ void MatroskaExtractor::addTracks() { } else if (!strcmp("A_VORBIS", codecID)) { meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS); - addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize); + err = addVorbisCodecInfo( + meta, codecPrivate, codecPrivateSize); } else if (!strcmp("A_MPEG/L3", codecID)) { meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG); } else { @@ -872,6 +913,11 @@ void MatroskaExtractor::addTracks() { continue; } + if (err != OK) { + ALOGE("skipping track, codec specific data was malformed."); + continue; + } + long long durationNs = mSegment->GetDuration(); meta->setInt64(kKeyDuration, (durationNs + 500) / 1000); diff --git a/media/libstagefright/mp4/FragmentedMP4Parser.cpp b/media/libstagefright/mp4/FragmentedMP4Parser.cpp index 7fe4e63..54c3d63 100644 --- a/media/libstagefright/mp4/FragmentedMP4Parser.cpp +++ b/media/libstagefright/mp4/FragmentedMP4Parser.cpp @@ -1971,8 +1971,8 @@ status_t FragmentedMP4Parser::parseTrackFragmentRun( } void FragmentedMP4Parser::copyBuffer( - sp<ABuffer> *dst, size_t offset, uint64_t size, size_t extra) const { - sp<ABuffer> buf = new ABuffer(size + extra); + sp<ABuffer> *dst, size_t offset, uint64_t size) const { + sp<ABuffer> buf = new ABuffer(size); memcpy(buf->data(), mBuffer->data() + offset, size); *dst = buf; diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp index 3747b3b..6e1c04d 100644 --- a/media/libstagefright/omx/SoftOMXPlugin.cpp +++ b/media/libstagefright/omx/SoftOMXPlugin.cpp @@ -53,6 +53,7 @@ static const struct { { "OMX.google.vpx.decoder", "vpxdec", "video_decoder.vpx" }, { "OMX.google.raw.decoder", "rawdec", "audio_decoder.raw" }, { "OMX.google.flac.encoder", "flacenc", "audio_encoder.flac" }, + { "OMX.google.gsm.decoder", "gsmdec", "audio_decoder.gsm" }, }; static const size_t kNumComponents = diff --git a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp index bc35aef..b913124 100644 --- a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp +++ b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp @@ -271,6 +271,7 @@ sp<ABuffer> TunnelRenderer::dequeueBuffer() { if (mFirstFailedAttemptUs + 50000ll > ALooper::GetNowUs()) { // We're willing to wait a little while to get the right packet. +#if 0 if (!mRequestedRetransmission) { ALOGI("requesting retransmission of seqNo %d", (mLastDequeuedExtSeqNo + 1) & 0xffff); @@ -280,7 +281,9 @@ sp<ABuffer> TunnelRenderer::dequeueBuffer() { notify->post(); mRequestedRetransmission = true; - } else { + } else +#endif + { ALOGI("still waiting for the correct packet to arrive."); } diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp index fcd20d4..c3e0470 100644 --- a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp +++ b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp @@ -475,9 +475,10 @@ void WifiDisplaySink::onGetParameterRequest( int32_t cseq, const sp<ParsedMessage> &data) { AString body = - "wfd_video_formats: xxx\r\n" - "wfd_audio_codecs: xxx\r\n" - "wfd_client_rtp_ports: RTP/AVP/UDP;unicast xxx 0 mode=play\r\n"; + "wfd_video_formats: " + "28 00 02 02 FFFFFFFF 0000000 00000000 00 0000 0000 00 none none\r\n" + "wfd_audio_codecs: AAC 0000000F 00\r\n" + "wfd_client_rtp_ports: RTP/AVP/UDP;unicast 19000 0 mode=play\r\n"; AString response = "RTSP/1.0 200 OK\r\n"; AppendCommonResponse(&response, cseq); diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk index bd9421c..4416b52 100644 --- a/services/audioflinger/Android.mk +++ b/services/audioflinger/Android.mk @@ -19,11 +19,9 @@ LOCAL_SRC_FILES:= \ AudioResampler.cpp.arm \ AudioPolicyService.cpp \ ServiceUtilities.cpp \ + AudioResamplerCubic.cpp.arm \ AudioResamplerSinc.cpp.arm -# uncomment to enable AudioResampler::MED_QUALITY -# LOCAL_SRC_FILES += AudioResamplerCubic.cpp.arm - LOCAL_SRC_FILES += StateQueue.cpp # uncomment for debugging timing problems related to StateQueue::push() @@ -33,7 +31,6 @@ LOCAL_C_INCLUDES := \ $(call include-path-for, audio-effects) \ $(call include-path-for, audio-utils) -# FIXME keep libmedia_native but remove libmedia after split LOCAL_SHARED_LIBRARIES := \ libaudioutils \ libcommon_time_client \ @@ -41,7 +38,6 @@ LOCAL_SHARED_LIBRARIES := \ libutils \ libbinder \ libmedia \ - libmedia_native \ libnbaio \ libhardware \ libhardware_legacy \ @@ -74,10 +70,37 @@ LOCAL_CFLAGS += -UFAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE # 47.5 seconds at 44.1 kHz, 8 megabytes # LOCAL_CFLAGS += -DTEE_SINK_FRAMES=0x200000 +# uncomment for dumpsys to write most recent audio input to .wav file +# 47.5 seconds at 44.1 kHz, 8 megabytes +# LOCAL_CFLAGS += -DTEE_SINK_INPUT_FRAMES=0x200000 + # uncomment to enable the audio watchdog # LOCAL_SRC_FILES += AudioWatchdog.cpp # LOCAL_CFLAGS += -DAUDIO_WATCHDOG include $(BUILD_SHARED_LIBRARY) +# +# build audio resampler test tool +# +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + test-resample.cpp \ + AudioResampler.cpp.arm \ + AudioResamplerCubic.cpp.arm \ + AudioResamplerSinc.cpp.arm + +LOCAL_SHARED_LIBRARIES := \ + libdl \ + libcutils \ + libutils + +LOCAL_MODULE:= test-resample + +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + + include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 76d6447..384f268 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -417,6 +417,12 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args) audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); dev->dump(dev, fd); } + + // dump the serially shared record tee sink + if (mRecordTeeSource != 0) { + dumpTee(fd, mRecordTeeSource); + } + if (locked) mLock.unlock(); } return NO_ERROR; @@ -444,8 +450,8 @@ sp<IAudioTrack> AudioFlinger::createTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, - IAudioFlinger::track_flags_t flags, + size_t frameCount, + IAudioFlinger::track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, @@ -466,6 +472,14 @@ sp<IAudioTrack> AudioFlinger::createTrack( goto Exit; } + // client is responsible for conversion of 8-bit PCM to 16-bit PCM, + // and we don't yet support 8.24 or 32-bit PCM + if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { + ALOGE("createTrack() invalid format %d", format); + lStatus = BAD_VALUE; + goto Exit; + } + { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); @@ -941,8 +955,8 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const { -// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", -// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); + ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", + ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); @@ -1112,7 +1126,8 @@ void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, c // removeClient_l() must be called with AudioFlinger::mLock held void AudioFlinger::removeClient_l(pid_t pid) { - ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); + ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), + IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } @@ -1215,7 +1230,8 @@ void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) { IoConfigEvent *ioEvent = new IoConfigEvent(event, param); mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); - ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); + ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, + param); mWaitWorkCV.signal(); } @@ -1244,7 +1260,8 @@ void AudioFlinger::ThreadBase::processConfigEvents() PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " + "error %d", prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); } } break; @@ -1282,7 +1299,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "standby: %d\n", mStandby); result.append(buffer); - snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); + snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); @@ -1661,7 +1678,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); - snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", + ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); result.append(buffer); @@ -1712,20 +1730,20 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, - IAudioFlinger::track_flags_t flags, + IAudioFlinger::track_flags_t *flags, pid_t tid, status_t *status) { sp<Track> track; status_t lStatus; - bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; + bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; // client expresses a preference for FAST, but we get the final say - if (flags & IAudioFlinger::TRACK_FAST) { + if (*flags & IAudioFlinger::TRACK_FAST) { if ( // not timed (!isTimed) && @@ -1766,12 +1784,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac frameCount, mFrameCount); } else { ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " - "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " + "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); - flags &= ~IAudioFlinger::TRACK_FAST; + *flags &= ~IAudioFlinger::TRACK_FAST; // For compatibility with AudioTrack calculation, buffer depth is forced // to be at least 2 x the normal mixer frame count and cover audio hardware latency. // This is probably too conservative, but legacy application code may depend on it. @@ -1781,7 +1799,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (minBufCount < 2) { minBufCount = 2; } - int minFrameCount = mNormalFrameCount * minBufCount; + size_t minFrameCount = mNormalFrameCount * minBufCount; if (frameCount < minFrameCount) { frameCount = minFrameCount; } @@ -1791,7 +1809,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (mType == DIRECT) { if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { - ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" + ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " "for output %p with format %d", sampleRate, format, channelMask, mOutput, mFormat); lStatus = BAD_VALUE; @@ -1801,7 +1819,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac } else { // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (sampleRate > mSampleRate*2) { - ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); + ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); lStatus = BAD_VALUE; goto Exit; } @@ -1835,7 +1853,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (!isTimed) { track = new Track(this, client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, sessionId, flags); + channelMask, frameCount, sharedBuffer, sessionId, *flags); } else { track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId); @@ -1854,7 +1872,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac chain->incTrackCnt(); } - if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { + if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { pid_t callingPid = IPCThreadState::self()->getCallingPid(); // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, // so ask activity manager to do this on our behalf @@ -1959,7 +1977,8 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) if (track->mainBuffer() != mMixBuffer) { sp<EffectChain> chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); + ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), + track->sessionId()); chain->incActiveTrackCnt(); } } @@ -2025,7 +2044,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = NULL; - ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); + ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, + param); switch (event) { case AudioSystem::OUTPUT_OPENED: @@ -2033,7 +2053,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { desc.channels = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; - desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) + desc.frameCount = mNormalFrameCount; // FIXME see + // AudioFlinger::frameCount(audio_io_handle_t) desc.latency = latency(); param2 = &desc; break; @@ -2062,7 +2083,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() // Calculate size of normal mix buffer relative to the HAL output buffer size double multiplier = 1.0; - if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { + if (mType == MIXER && (kUseFastMixer == FastMixer_Static || + kUseFastMixer == FastMixer_Dynamic)) { size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer @@ -2081,9 +2103,10 @@ void AudioFlinger::PlaybackThread::readOutputParameters() multiplier = (double) maxNormalFrameCount / (double) mFrameCount; } } else { - // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC - // (it would be unusual for the normal mix buffer size to not be a multiple of fast - // track, but we sometimes have to do this to satisfy the maximum frame count constraint) + // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL + // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast + // track, but we sometimes have to do this to satisfy the maximum frame count + // constraint) // FIXME this rounding up should not be done if no HAL SRC uint32_t truncMult = (uint32_t) multiplier; if ((truncMult & 1)) { @@ -2097,7 +2120,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mNormalFrameCount = multiplier * mFrameCount; // round up to nearest 16 frames to satisfy AudioMixer mNormalFrameCount = (mNormalFrameCount + 15) & ~15; - ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); + ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, + mNormalFrameCount); delete[] mMixBuffer; mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; @@ -2151,7 +2175,7 @@ uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const for (size_t i = 0; i < mTracks.size(); ++i) { sp<Track> track = mTracks[i]; if (sessionId == track->sessionId() && - !(track->mCblk->flags & CBLK_INVALID_MSK)) { + !(track->mCblk->flags & CBLK_INVALID)) { result |= TRACK_SESSION; break; } @@ -2170,7 +2194,7 @@ uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) for (size_t i = 0; i < mTracks.size(); i++) { sp<Track> track = mTracks[i]; if (sessionId == track->sessionId() && - !(track->mCblk->flags & CBLK_INVALID_MSK)) { + !(track->mCblk->flags & CBLK_INVALID)) { return AudioSystem::getStrategyForStream(track->streamType()); } } @@ -2235,7 +2259,8 @@ bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; } -void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) +void AudioFlinger::PlaybackThread::threadLoop_removeTracks( + const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); if (CC_UNLIKELY(count)) { @@ -2263,7 +2288,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud // mNormalSink below { ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); - ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " + ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " "mFrameCount=%d, mNormalFrameCount=%d", mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, mNormalFrameCount); @@ -2891,7 +2916,8 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { memset (mMixBuffer, 0, mixBufferSize); sleepTime = 0; - ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); + ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), + "anticipated start"); } // TODO add standby time extension fct of effect tail } @@ -3014,7 +3040,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } // indicate to client process that the track was disabled because of underrun; // it will then automatically call start() when data is available - android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); + android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); // remove from active list, but state remains ACTIVE [confusing but true] isActive = false; break; @@ -3108,7 +3134,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac uint32_t minFrames = 1; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { - if (t->sampleRate() == (int)mSampleRate) { + if (t->sampleRate() == mSampleRate) { minFrames = mNormalFrameCount; } else { // +1 for rounding and +1 for additional sample needed for interpolation @@ -3125,7 +3151,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { - //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, + this); mixedTracks++; @@ -3138,7 +3165,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if (chain != 0) { tracksWithEffect++; } else { - ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", + ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " + "session %d", name, track->sessionId()); } } @@ -3268,7 +3296,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac chain->clearInputBuffer(); } - //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, + cblk->server, this); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3293,7 +3322,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac tracksToRemove->add(track); // indicate to client process that the track was disabled because of underrun; // it will then automatically call start() when data is available - android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); + android_atomic_or(CBLK_DISABLED, &cblk->flags); // If one track is not ready, mark the mixer also not ready if: // - the mixer was ready during previous round OR // - no other track is ready @@ -3362,7 +3391,8 @@ track_is_ready: ; if (track->mainBuffer() != mMixBuffer) { chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); + ALOGV("stopping track on chain %p for session Id: %d", chain.get(), + track->sessionId()); chain->decActiveTrackCnt(); } } @@ -3375,7 +3405,8 @@ track_is_ready: ; // mix buffer must be cleared if all tracks are connected to an // effect chain as in this case the mixer will not write to // mix buffer and track effects will accumulate into it - if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { + if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || + (mixedTracks == 0 && fastTracks > 0)) { // FIXME as a performance optimization, should remember previous zero status memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); } @@ -3424,7 +3455,7 @@ void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamTy for (size_t i = 0; i < size; i++) { sp<Track> t = mTracks[i]; if (t->streamType() == streamType) { - android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); + android_atomic_or(CBLK_INVALID, &t->mCblk->flags); t->mCblk->cv.signal(); } } @@ -3580,39 +3611,18 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() return reconfig; } -void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) +void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) { - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - PlaybackThread::dumpInternals(fd, args); - - snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); - result.append(buffer); - write(fd, result.string(), result.size()); - - // Make a non-atomic copy of fast mixer dump state so it won't change underneath us - FastMixerDumpState copy = mFastMixerDumpState; - copy.dump(fd); - -#ifdef STATE_QUEUE_DUMP - // Similar for state queue - StateQueueObserverDump observerCopy = mStateQueueObserverDump; - observerCopy.dump(fd); - StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; - mutatorCopy.dump(fd); -#endif - - // Write the tee output to a .wav file - NBAIO_Source *teeSource = mTeeSource.get(); + NBAIO_Source *teeSource = source.get(); if (teeSource != NULL) { - char teePath[64]; + char teeTime[16]; struct timeval tv; gettimeofday(&tv, NULL); struct tm tm; localtime_r(&tv.tv_sec, &tm); - strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); + strftime(teeTime, sizeof(teeTime), "%T", &tm); + char teePath[64]; + sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); if (teeFd >= 0) { char wavHeader[44]; @@ -3622,7 +3632,7 @@ void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& ar NBAIO_Format format = teeSource->format(); unsigned channelCount = Format_channelCount(format); ALOG_ASSERT(channelCount <= FCC_2); - unsigned sampleRate = Format_sampleRate(format); + uint32_t sampleRate = Format_sampleRate(format); wavHeader[22] = channelCount; // number of channels wavHeader[24] = sampleRate; // sample rate wavHeader[25] = sampleRate >> 8; @@ -3660,6 +3670,34 @@ void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& ar fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); } } +} + +void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + PlaybackThread::dumpInternals(fd, args); + + snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); + result.append(buffer); + write(fd, result.string(), result.size()); + + // Make a non-atomic copy of fast mixer dump state so it won't change underneath us + FastMixerDumpState copy = mFastMixerDumpState; + copy.dump(fd); + +#ifdef STATE_QUEUE_DUMP + // Similar for state queue + StateQueueObserverDump observerCopy = mStateQueueObserverDump; + observerCopy.dump(fd); + StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; + mutatorCopy.dump(fd); +#endif + + // Write the tee output to a .wav file + dumpTee(fd, mTeeSource, mId); #ifdef AUDIO_WATCHDOG if (mAudioWatchdog != 0) { @@ -3731,7 +3769,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { - //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); + ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; @@ -3792,7 +3830,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep mEffectChains[0]->clearInputBuffer(); } - //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3982,7 +4020,8 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l() AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) - : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING), + : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), + DUPLICATING), mWaitTimeMs(UINT_MAX) { addOutputTrack(mainThread); @@ -4058,7 +4097,7 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) { Mutex::Autolock _l(mLock); // FIXME explain this formula - int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); + size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); OutputTrack *outputTrack = new OutputTrack(thread, this, mSampleRate, @@ -4103,18 +4142,21 @@ void AudioFlinger::DuplicatingThread::updateWaitTime_l() } -bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) +bool AudioFlinger::DuplicatingThread::outputsReady( + const SortedVector< sp<OutputTrack> > &outputTracks) { for (size_t i = 0; i < outputTracks.size(); i++) { sp<ThreadBase> thread = outputTracks[i]->thread().promote(); if (thread == 0) { - ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); + ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", + outputTracks[i].get()); return false; } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); // see note at standby() declaration if (playbackThread->standby() && !playbackThread->isSuspended()) { - ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); + ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), + thread.get()); return false; } } @@ -4143,7 +4185,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId) : RefBase(), @@ -4152,52 +4194,35 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( mCblk(NULL), // mBuffer // mBufferEnd - mFrameCount(0), + mStepCount(0), mState(IDLE), mSampleRate(sampleRate), mFormat(format), + mChannelMask(channelMask), + mChannelCount(popcount(channelMask)), + mFrameSize(audio_is_linear_pcm(format) ? + mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), mStepServerFailed(false), mSessionId(sessionId) - // mChannelCount - // mChannelMask { - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + // client == 0 implies sharedBuffer == 0 + ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); + + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); - uint8_t channelCount = popcount(channelMask); - size_t bufferSize = frameCount*channelCount*sizeof(int16_t); + size_t bufferSize = frameCount * mFrameSize; if (sharedBuffer == 0) { size += bufferSize; } - if (client != NULL) { + if (client != 0) { mCblkMemory = client->heap()->allocate(size); if (mCblkMemory != 0) { mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); - if (mCblk != NULL) { // construct the shared structure in-place. - new(mCblk) audio_track_cblk_t(); - // clear all buffers - mCblk->frameCount = frameCount; - mCblk->sampleRate = sampleRate; -// uncomment the following lines to quickly test 32-bit wraparound -// mCblk->user = 0xffff0000; -// mCblk->server = 0xffff0000; -// mCblk->userBase = 0xffff0000; -// mCblk->serverBase = 0xffff0000; - mChannelCount = channelCount; - mChannelMask = channelMask; - if (sharedBuffer == 0) { - mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); - memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer (other flags are cleared) - mCblk->flags = CBLK_UNDERRUN_ON; - } else { - mBuffer = sharedBuffer->pointer(); - } - mBufferEnd = (uint8_t *)mBuffer + bufferSize; - } + // can't assume mCblk != NULL } else { ALOGE("not enough memory for AudioTrack size=%u", size); client->heap()->dump("AudioTrack"); @@ -4205,23 +4230,29 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( } } else { mCblk = (audio_track_cblk_t *)(new uint8_t[size]); - // construct the shared structure in-place. + // assume mCblk != NULL + } + + // construct the shared structure in-place. + if (mCblk != NULL) { new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount = frameCount; mCblk->sampleRate = sampleRate; // uncomment the following lines to quickly test 32-bit wraparound -// mCblk->user = 0xffff0000; -// mCblk->server = 0xffff0000; -// mCblk->userBase = 0xffff0000; -// mCblk->serverBase = 0xffff0000; - mChannelCount = channelCount; - mChannelMask = channelMask; - mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); - memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer (other flags are cleared) - mCblk->flags = CBLK_UNDERRUN_ON; +// mCblk->user = 0xffff0000; +// mCblk->server = 0xffff0000; +// mCblk->userBase = 0xffff0000; +// mCblk->serverBase = 0xffff0000; + if (sharedBuffer == 0) { + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, bufferSize); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer (other flags are cleared) + mCblk->flags = CBLK_UNDERRUN; + } else { + mBuffer = sharedBuffer->pointer(); + } mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } @@ -4252,7 +4283,7 @@ AudioFlinger::ThreadBase::TrackBase::~TrackBase() void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { buffer->raw = NULL; - mFrameCount = buffer->frameCount; + mStepCount = buffer->frameCount; // FIXME See note at getNextBuffer() (void) step(); // ignore return value of step() buffer->frameCount = 0; @@ -4262,7 +4293,7 @@ bool AudioFlinger::ThreadBase::TrackBase::step() { bool result; audio_track_cblk_t* cblk = this->cblk(); - result = cblk->stepServer(mFrameCount); + result = cblk->stepServer(mStepCount, isOut()); if (!result) { ALOGV("stepServer failed acquiring cblk mutex"); mStepServerFailed = true; @@ -4281,23 +4312,22 @@ void AudioFlinger::ThreadBase::TrackBase::reset() { ALOGV("TrackBase::reset"); } -int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { - return (int)mCblk->sampleRate; +uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { + return mCblk->sampleRate; } void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { audio_track_cblk_t* cblk = this->cblk(); - size_t frameSize = cblk->frameSize; - int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; - int8_t *bufferEnd = bufferStart + frames * frameSize; + int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; + int8_t *bufferEnd = bufferStart + frames * mFrameSize; // Check validity of returned pointer in case the track control block would have been corrupted. ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), "TrackBase::getBuffer buffer out of range:\n" " start: %p, end %p , mBuffer %p mBufferEnd %p\n" - " server %u, serverBase %u, user %u, userBase %u, frameSize %d", + " server %u, serverBase %u, user %u, userBase %u, frameSize %u", bufferStart, bufferEnd, mBuffer, mBufferEnd, - cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); + cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); return bufferStart; } @@ -4318,11 +4348,12 @@ AudioFlinger::PlaybackThread::Track::Track( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags) - : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), + : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, + sessionId), mMute(false), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed @@ -4339,9 +4370,6 @@ AudioFlinger::PlaybackThread::Track::Track( mCachedVolume(1.0) { if (mCblk != NULL) { - // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of - // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack - mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); // to avoid leaking a track name, do not allocate one unless there is an mCblk mName = thread->getTrackName_l(channelMask, sessionId); mCblk->mName = mName; @@ -4351,7 +4379,6 @@ AudioFlinger::PlaybackThread::Track::Track( } // only allocate a fast track index if we were able to allocate a normal track name if (flags & IAudioFlinger::TRACK_FAST) { - mCblk->flags |= CBLK_FAST; // atomic op not needed yet ALOG_ASSERT(thread->mFastTrackAvailMask != 0); int i = __builtin_ctz(thread->mFastTrackAvailMask); ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); @@ -4366,7 +4393,8 @@ AudioFlinger::PlaybackThread::Track::Track( thread->mFastTrackAvailMask &= ~(1 << i); } } - ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); + ALOGV("Track constructor name %d, calling pid %d", mName, + IPCThreadState::self()->getCallingPid()); } AudioFlinger::PlaybackThread::Track::~Track() @@ -4408,8 +4436,8 @@ void AudioFlinger::PlaybackThread::Track::destroy() /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { - result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " - " Server User Main buf Aux Buf Flags Underruns\n"); + result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " + "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) @@ -4479,7 +4507,7 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) mFormat, mChannelMask, mSessionId, - mFrameCount, + mStepCount, mCblk->frameCount, stateChar, mMute, @@ -4519,7 +4547,7 @@ status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( } // FIXME Same as above - framesReady = cblk->framesReady(); + framesReady = cblk->framesReadyOut(); if (CC_LIKELY(framesReady)) { uint32_t s = cblk->server; @@ -4554,7 +4582,7 @@ getNextBuffer_exit: // the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. // FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. size_t AudioFlinger::PlaybackThread::Track::framesReady() const { - return mCblk->framesReady(); + return mCblk->framesReadyOut(); } // Don't call for fast tracks; the framesReady() could result in priority inversion @@ -4562,9 +4590,9 @@ bool AudioFlinger::PlaybackThread::Track::isReady() const { if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; if (framesReady() >= mCblk->frameCount || - (mCblk->flags & CBLK_FORCEREADY_MSK)) { + (mCblk->flags & CBLK_FORCEREADY)) { mFillingUpStatus = FS_FILLED; - android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); return true; } return false; @@ -4636,7 +4664,8 @@ void AudioFlinger::PlaybackThread::Track::stop() // and then to STOPPED and reset() when presentation is complete mState = STOPPING_1; } - ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); + ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, + playbackThread); } if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { thread->mLock.unlock(); @@ -4681,7 +4710,7 @@ void AudioFlinger::PlaybackThread::Track::flush() if (thread != 0) { Mutex::Autolock _l(thread->mLock); if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && - mState != PAUSING) { + mState != PAUSING && mState != IDLE && mState != FLUSHED) { return; } // No point remaining in PAUSED state after a flush => go to @@ -4706,8 +4735,8 @@ void AudioFlinger::PlaybackThread::Track::reset() TrackBase::reset(); // Force underrun condition to avoid false underrun callback until first data is // written to buffer - android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); - android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); mFillingUpStatus = FS_FILLING; mResetDone = true; if (mState == FLUSHED) { @@ -4848,6 +4877,11 @@ status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& return NO_ERROR; } +bool AudioFlinger::PlaybackThread::Track::isOut() const +{ + return true; +} + // timed audio tracks sp<AudioFlinger::PlaybackThread::TimedTrack> @@ -4858,7 +4892,7 @@ AudioFlinger::PlaybackThread::TimedTrack::create( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId) { if (!client->reserveTimedTrack()) @@ -4876,7 +4910,7 @@ AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId) : Track(thread, client, streamType, sampleRate, format, channelMask, @@ -4982,7 +5016,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { // this frame in media time units and adding it to the PTS of the // buffer. int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() - / mCblk->frameSize; + / mFrameSize; if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, &bufEnd)) { @@ -5042,7 +5076,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( " bytes. (update reason: \"%s\")", bufBytes, consumedAlready, logTag); - uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; + uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, "Bad bookkeeping while updating frames pending. Should have at" " least %u queued frames, but we think we have only %u. (update" @@ -5063,7 +5097,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( Mutex::Autolock _l(mTimedBufferQueueLock); - uint32_t bufFrames = buffer->size() / mCblk->frameSize; + uint32_t bufFrames = buffer->size() / mFrameSize; mFramesPendingInQueue += bufFrames; mTimedBufferQueue.add(TimedBuffer(buffer, pts)); @@ -5160,7 +5194,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( // adjust the head buffer's PTS to reflect the portion of the head buffer // that has already been consumed int64_t effectivePTS = headLocalPTS + - ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); + ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); // Calculate the delta in samples between the head of the input buffer // queue and the start of the next output buffer that will be written. @@ -5225,7 +5259,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( // the next input sample is late uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); size_t onTimeSamplePosition = - head.position() + lateFrames * mCblk->frameSize; + head.position() + lateFrames * mFrameSize; if (onTimeSamplePosition > head.buffer()->size()) { // all the remaining samples in the head are too late, so @@ -5260,7 +5294,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( head.position()); uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / - mCblk->frameSize); + mFrameSize); size_t framesRequested = buffer->frameCount; buffer->frameCount = min(framesLeftInHead, framesRequested); @@ -5275,9 +5309,9 @@ void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { // lazily allocate a buffer filled with silence - if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { + if (mTimedSilenceBufferSize < numFrames * mFrameSize) { delete [] mTimedSilenceBuffer; - mTimedSilenceBufferSize = numFrames * mCblk->frameSize; + mTimedSilenceBufferSize = numFrames * mFrameSize; mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); } @@ -5325,7 +5359,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( start, end, buffer->raw); head.setPosition(head.position() + - (buffer->frameCount * mCblk->frameSize)); + (buffer->frameCount * mFrameSize)); mQueueHeadInFlight = false; ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, @@ -5371,22 +5405,13 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, int sessionId) : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), mOverflow(false) { - if (mCblk != NULL) { - ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); - if (format == AUDIO_FORMAT_PCM_16_BIT) { - mCblk->frameSize = mChannelCount * sizeof(int16_t); - } else if (format == AUDIO_FORMAT_PCM_8_BIT) { - mCblk->frameSize = mChannelCount * sizeof(int8_t); - } else { - mCblk->frameSize = sizeof(int8_t); - } - } + ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); } AudioFlinger::RecordThread::RecordTrack::~RecordTrack() @@ -5395,7 +5420,8 @@ AudioFlinger::RecordThread::RecordTrack::~RecordTrack() } // AudioBufferProvider interface -status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesAvail; @@ -5408,7 +5434,8 @@ status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvi mStepServerFailed = false; } - framesAvail = cblk->framesAvailable_l(); + // FIXME lock is not actually held, so overrun is possible + framesAvail = cblk->framesAvailableIn_l(); if (CC_LIKELY(framesAvail)) { uint32_t s = cblk->server; @@ -5455,7 +5482,7 @@ void AudioFlinger::RecordThread::RecordTrack::stop() TrackBase::reset(); // Force overrun condition to avoid false overrun callback until first data is // read from buffer - android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); } recordThread->mLock.unlock(); if (doStop) { @@ -5466,7 +5493,7 @@ void AudioFlinger::RecordThread::RecordTrack::stop() /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) { - result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User FrameCount\n"); + result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); } void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) @@ -5476,7 +5503,7 @@ void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) mFormat, mChannelMask, mSessionId, - mFrameCount, + mStepCount, mState, mCblk->sampleRate, mCblk->server, @@ -5484,6 +5511,10 @@ void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) mCblk->frameCount); } +bool AudioFlinger::RecordThread::RecordTrack::isOut() const +{ + return false; +} // ---------------------------------------------------------------------------- @@ -5493,19 +5524,18 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount) + size_t frameCount) : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0, IAudioFlinger::TRACK_DEFAULT), - mActive(false), mSourceThread(sourceThread) + mActive(false), mSourceThread(sourceThread), mBuffers(NULL) { if (mCblk != NULL) { - mCblk->flags |= CBLK_DIRECTION_OUT; - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); mOutBuffer.frameCount = 0; playbackThread->mTracks.add(this); ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ - "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", + "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); } else { @@ -5587,7 +5617,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { - ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); + ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, + mThread.unsafe_get()); outputBufferFull = true; break; } @@ -5599,9 +5630,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr } } - uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; + uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : + pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); - mCblk->stepUser(outFrames); + mCblk->stepUserOut(outFrames); pInBuffer->frameCount -= outFrames; pInBuffer->i16 += outFrames * channelCount; mOutBuffer.frameCount -= outFrames; @@ -5612,7 +5644,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; - ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { break; } @@ -5628,11 +5661,14 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * + sizeof(int16_t)); mBufferQueue.add(pInBuffer); - ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { - ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); + ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", + mThread.unsafe_get(), this); } } } @@ -5657,17 +5693,18 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr return outputBufferFull; } -status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) +status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( + AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { int active; status_t result; audio_track_cblk_t* cblk = mCblk; uint32_t framesReq = buffer->frameCount; -// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); + ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); buffer->frameCount = 0; - uint32_t framesAvail = cblk->framesAvailable(); + uint32_t framesAvail = cblk->framesAvailableOut(); if (framesAvail == 0) { @@ -5685,7 +5722,7 @@ status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProv } // read the server count again start_loop_here: - framesAvail = cblk->framesAvailable_l(); + framesAvail = cblk->framesAvailableOut_l(); } } @@ -5705,7 +5742,7 @@ status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProv } buffer->frameCount = framesReq; - buffer->raw = (void *)cblk->buffer(u); + buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); return NO_ERROR; } @@ -5881,7 +5918,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, IAudioFlinger::track_flags_t flags, pid_t tid, int *sessionId, @@ -5921,13 +5958,14 @@ sp<IAudioRecord> AudioFlinger::openRecord( *sessionId = lSessionId; } } - // create new record track. The record track uses one track in mHardwareMixerThread by convention. + // create new record track. + // The record track uses one track in mHardwareMixerThread by convention. recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); } if (lStatus != NO_ERROR) { - // remove local strong reference to Client before deleting the RecordTrack so that the Client - // destructor is called by the TrackBase destructor with mLock held + // remove local strong reference to Client before deleting the RecordTrack so that the + // Client destructor is called by the TrackBase destructor with mLock held client.clear(); recordTrack.clear(); goto Exit; @@ -5946,7 +5984,8 @@ Exit: // ---------------------------------------------------------------------------- -AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { @@ -5961,7 +6000,8 @@ sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } -status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { +status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, + int triggerSession) { ALOGV("RecordHandle::start()"); return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); } @@ -5988,18 +6028,21 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t id, - audio_devices_t device) : + audio_devices_t device, + const sp<NBAIO_Sink>& teeSink) : ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), // mRsmpInIndex and mInputBytes set by readInputParameters() mReqChannelCount(popcount(channelMask)), - mReqSampleRate(sampleRate) + mReqSampleRate(sampleRate), // mBytesRead is only meaningful while active, and so is cleared in start() // (but might be better to also clear here for dump?) + mTeeSink(teeSink) { snprintf(mName, kNameLength, "AudioIn_%X", id); readInputParameters(); + } @@ -6106,7 +6149,8 @@ bool AudioFlinger::RecordThread::threadLoop() size_t framesIn = mFrameCount - mRsmpInIndex; if (framesIn) { int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + mActiveTrack->mFrameSize; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; @@ -6125,14 +6169,17 @@ bool AudioFlinger::RecordThread::threadLoop() } } if (framesOut && mFrameCount == mRsmpInIndex) { + void *readInto; if (framesOut == mFrameCount && - ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { - mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); + ((int)mChannelCount == mReqChannelCount || + mFormat != AUDIO_FORMAT_PCM_16_BIT)) { + readInto = buffer.raw; framesOut = 0; } else { - mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); + readInto = mRsmpInBuffer; mRsmpInIndex = 0; } + mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { @@ -6145,6 +6192,9 @@ bool AudioFlinger::RecordThread::threadLoop() mRsmpInIndex = mFrameCount; framesOut = 0; buffer.frameCount = 0; + } else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); } } } @@ -6156,12 +6206,14 @@ bool AudioFlinger::RecordThread::threadLoop() if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; } - mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() - // are 32 bit aligned which should be always true. + mResampler->resample(mRsmpOutBuffer, framesOut, + this /* AudioBufferProvider* */); + // ditherAndClamp() works as long as all buffers returned by + // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. if (mChannelCount == 2 && mReqChannelCount == 1) { ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: do post stereo to mono conversion + // the resampler always outputs stereo samples: + // do post stereo to mono conversion downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, framesOut); } else { @@ -6243,7 +6295,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, @@ -6500,7 +6552,7 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a result.append(buffer); snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); result.append(buffer); - snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); + snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); result.append(buffer); } else { result.append("No active record client\n"); @@ -6595,7 +6647,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() AudioParameter param = AudioParameter(keyValuePair); int value; audio_format_t reqFormat = mFormat; - int reqSamplingRate = mReqSampleRate; + uint32_t reqSamplingRate = mReqSampleRate; int reqChannelCount = mReqChannelCount; if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { @@ -6635,7 +6687,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() status = BAD_VALUE; } else { mInDevice = value; - // disable AEC and NS if the device is a BT SCO headset supporting those pre processings + // disable AEC and NS if the device is a BT SCO headset supporting those + // pre processings if (mTracks.size() > 0) { bool suspend = audio_is_bluetooth_sco_device(mInDevice) && mAudioFlinger->btNrecIsOff(); @@ -6657,7 +6710,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() mAudioSource = (audio_source_t)value; } if (status == NO_ERROR) { - status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); + status = mInput->stream->common.set_parameters(&mInput->stream->common, + keyValuePair.string()); if (status == INVALID_OPERATION) { inputStandBy(); status = mInput->stream->common.set_parameters(&mInput->stream->common, @@ -6667,8 +6721,10 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() if (status == BAD_VALUE && reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && reqFormat == AUDIO_FORMAT_PCM_16_BIT && - ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && - popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && + ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) + <= (2 * reqSamplingRate)) && + popcount(mInput->stream->common.get_channels(&mInput->stream->common)) + <= FCC_2 && (reqChannelCount <= FCC_2)) { status = NO_ERROR; } @@ -6762,7 +6818,8 @@ void AudioFlinger::RecordThread::readInputParameters() mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); mRsmpOutBuffer = new int32_t[mFrameCount * 2]; - // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples + // optmization: if mono to mono, alter input frame count as if we were inputing + // stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { mFrameCount >>= 1; } @@ -6924,14 +6981,14 @@ audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) // ---------------------------------------------------------------------------- -int32_t AudioFlinger::getPrimaryOutputSamplingRate() +uint32_t AudioFlinger::getPrimaryOutputSamplingRate() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); return thread != NULL ? thread->sampleRate() : 0; } -int32_t AudioFlinger::getPrimaryOutputFrameCount() +size_t AudioFlinger::getPrimaryOutputFrameCount() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); @@ -6989,7 +7046,8 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, &outStream); mHardwareStatus = AUDIO_HW_IDLE; - ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " + "Channels %x, status %d", outStream, config.sample_rate, config.format, @@ -7042,7 +7100,8 @@ audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { - ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); + ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, + output2); return 0; } @@ -7077,7 +7136,8 @@ status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) if (thread->type() == ThreadBase::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { - DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); + DuplicatingThread *dupThread = + (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } @@ -7164,16 +7224,17 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); - ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " + "status %d", inStream, config.sample_rate, config.format, config.channel_mask, status); - // If the input could not be opened with the requested parameters and we can handle the conversion internally, - // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo - // or stereo to mono conversions on 16 bit PCM inputs. + // If the input could not be opened with the requested parameters and we can handle the + // conversion internally, try to open again with the proposed parameters. The AudioFlinger can + // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && @@ -7184,18 +7245,66 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, } if (status == NO_ERROR && inStream != NULL) { + + // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, + // or (re-)create if current Pipe is idle and does not match the new format + sp<NBAIO_Sink> teeSink; +#ifdef TEE_SINK_INPUT_FRAMES + enum { + TEE_SINK_NO, // don't copy input + TEE_SINK_NEW, // copy input using a new pipe + TEE_SINK_OLD, // copy input using an existing pipe + } kind; + NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), + popcount(inStream->common.get_channels(&inStream->common))); + if (format == Format_Invalid) { + kind = TEE_SINK_NO; + } else if (mRecordTeeSink == 0) { + kind = TEE_SINK_NEW; + } else if (mRecordTeeSink->getStrongCount() != 1) { + kind = TEE_SINK_NO; + } else if (format == mRecordTeeSink->format()) { + kind = TEE_SINK_OLD; + } else { + kind = TEE_SINK_NEW; + } + switch (kind) { + case TEE_SINK_NEW: { + Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); + size_t numCounterOffers = 0; + const NBAIO_Format offers[1] = {format}; + ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + PipeReader *pipeReader = new PipeReader(*pipe); + numCounterOffers = 0; + index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + mRecordTeeSink = pipe; + mRecordTeeSource = pipeReader; + teeSink = pipe; + } + break; + case TEE_SINK_OLD: + teeSink = mRecordTeeSink; + break; + case TEE_SINK_NO: + default: + break; + } +#endif AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); // Start record thread // RecorThread require both input and output device indication to forward to audio // pre processing modules audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); + thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id, - device); + device, teeSink); mRecordThreads.add(id, thread); ALOGV("openInput() created record thread: ID %d thread %p", id, thread); if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; @@ -8003,7 +8112,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c for (size_t i = 0; i < mTracks.size(); ++i) { sp<Track> track = mTracks[i]; if (session == track->sessionId()) { - ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); + ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), + buffer); track->setMainBuffer(buffer); chain->incTrackCnt(); } @@ -8594,7 +8704,7 @@ status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, void *pReplyData) { Mutex::Autolock _l(mLock); -// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); + ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); if (mState == DESTROYED || mEffectInterface == NULL) { return NO_INIT; @@ -8845,12 +8955,15 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) result.append("\t\tDescriptor:\n"); snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, - mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], + mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], + mDescriptor.uuid.node[2], mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, - mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], + mDescriptor.type.timeLow, mDescriptor.type.timeMid, + mDescriptor.type.timeHiAndVersion, + mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], + mDescriptor.type.node[2], mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", @@ -8934,7 +9047,8 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, mBuffer = (uint8_t *)mCblk + bufOffset; } } else { - ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); + ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + + sizeof(effect_param_cblk_t)); return; } } @@ -9049,8 +9163,8 @@ status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, uint32_t *replySize, void *pReplyData) { -// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", -// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); + ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", + cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); // only get parameter command is permitted for applications not controlling the effect if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { @@ -9061,8 +9175,9 @@ status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, // handle commands that are not forwarded transparently to effect engine if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { - // No need to trylock() here as this function is executed in the binder thread serving a particular client process: - // no risk to block the whole media server process or mixer threads is we are stuck here + // No need to trylock() here as this function is executed in the binder thread serving a + // particular client process: no risk to block the whole media server process or mixer + // threads if we are stuck here Mutex::Autolock _l(mCblk->lock); if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { @@ -9202,7 +9317,8 @@ AudioFlinger::EffectChain::~EffectChain() } // getEffectFromDesc_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( + effect_descriptor_t *descriptor) { size_t size = mEffects.size(); @@ -9361,7 +9477,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) // check invalid effect chaining combinations if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { - ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); + ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", + desc.name, d.name); return INVALID_OPERATION; } // remember position of first insert effect and by default @@ -9412,7 +9529,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) } mEffects.insertAt(effect, idx_insert); - ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); + ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, + idx_insert); } effect->configure(); return NO_ERROR; @@ -9443,7 +9561,8 @@ size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) } } mEffects.removeAt(i); - ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); + ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), + this, i); break; } } @@ -9667,7 +9786,8 @@ void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) for (size_t i = 0; i < types.size(); i++) { setEffectSuspended_l(types[i], false); } - ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); + ALOGV("setEffectSuspendedAll_l() remove entry for %08x", + mSuspendedEffects.keyAt(index)); mSuspendedEffects.removeItem((int)kKeyForSuspendAll); } } @@ -9693,7 +9813,8 @@ bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descript return true; } -void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) +void AudioFlinger::EffectChain::getSuspendEligibleEffects( + Vector< sp<AudioFlinger::EffectModule> > &effects) { effects.clear(); for (size_t i = 0; i < mEffects.size(); i++) { diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 49e2b2c..830dfe9 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -92,8 +92,8 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, - IAudioFlinger::track_flags_t flags, + size_t frameCount, + IAudioFlinger::track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, @@ -106,7 +106,7 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, IAudioFlinger::track_flags_t flags, pid_t tid, int *sessionId, @@ -207,8 +207,8 @@ public: virtual audio_module_handle_t loadHwModule(const char *name); - virtual int32_t getPrimaryOutputSamplingRate(); - virtual int32_t getPrimaryOutputFrameCount(); + virtual uint32_t getPrimaryOutputSamplingRate(); + virtual size_t getPrimaryOutputFrameCount(); virtual status_t onTransact( uint32_t code, @@ -269,12 +269,14 @@ private: virtual ~AudioFlinger(); // call in any IAudioFlinger method that accesses mPrimaryHardwareDev - status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } + status_t initCheck() const { return mPrimaryHardwareDev == NULL ? + NO_INIT : NO_ERROR; } // RefBase virtual void onFirstRef(); - AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); + AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, + audio_devices_t devices); void purgeStaleEffects_l(); // standby delay for MIXER and DUPLICATING playback threads is read from property @@ -356,7 +358,7 @@ private: RECORD // Thread class is RecordThread }; - ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, + ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, type_t type); virtual ~ThreadBase(); @@ -388,7 +390,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId); virtual ~TrackBase(); @@ -421,7 +423,7 @@ private: audio_channel_mask_t channelMask() const { return mChannelMask; } - int sampleRate() const; // FIXME inline after cblk sr moved + uint32_t sampleRate() const; // FIXME inline after cblk sr moved // Return a pointer to the start of a contiguous slice of the track buffer. // Parameter 'offset' is the requested start position, expressed in @@ -451,9 +453,12 @@ private: return mState == TERMINATED; } - bool step(); + bool step(); // mStepCount is an implicit input void reset(); + virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack, + // this could be a track type if needed later + const wp<ThreadBase> mThread; /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const sp<IMemory> mCblkMemory; @@ -461,16 +466,20 @@ private: void* mBuffer; // start of track buffer, typically in shared memory void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize // is based on mChannelCount and 16-bit samples - uint32_t mFrameCount; + uint32_t mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of + // time of releaseBuffer() for later use by step() // we don't really need a lock for these track_state mState; const uint32_t mSampleRate; // initial sample rate only; for tracks which // support dynamic rates, the current value is in control block const audio_format_t mFormat; + const audio_channel_mask_t mChannelMask; + const uint8_t mChannelCount; + const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, + // where for AudioTrack (but not AudioRecord), + // 8-bit PCM samples are stored as 16-bit bool mStepServerFailed; const int mSessionId; - uint8_t mChannelCount; - audio_channel_mask_t mChannelMask; Vector < sp<SyncEvent> >mSyncEvents; }; @@ -746,7 +755,8 @@ private: const sp<PMDeathRecipient> mDeathRecipient; // list of suspended effects per session and per type. The first vector is // keyed by session ID, the second by type UUID timeLow field - KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; + KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > + mSuspendedSessions; }; struct stream_type_t { @@ -780,7 +790,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags); @@ -788,7 +798,8 @@ private: static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); - virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); virtual void stop(); void pause(); @@ -810,6 +821,7 @@ private: // implement FastMixerState::VolumeProvider interface virtual uint32_t getVolumeLR(); + virtual status_t setSyncEvent(const sp<SyncEvent>& event); protected: @@ -822,7 +834,8 @@ private: Track& operator = (const Track&); // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); // releaseBuffer() not overridden virtual size_t framesReady() const; @@ -847,12 +860,16 @@ private: sp<IMemory> sharedBuffer() const { return mSharedBuffer; } + // framesWritten is cumulative, never reset, and is shared all tracks + // audioHalFrames is derived from output latency + // FIXME parameters not needed, could get them from the thread bool presentationComplete(size_t framesWritten, size_t audioHalFrames); public: void triggerEvents(AudioSystem::sync_event_t type); virtual bool isTimedTrack() const { return false; } bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } + virtual bool isOut() const; protected: @@ -876,8 +893,9 @@ private: int32_t *mAuxBuffer; int mAuxEffectId; bool mHasVolumeController; - size_t mPresentationCompleteFrames; // number of frames written to the audio HAL - // when this track will be fully rendered + size_t mPresentationCompleteFrames; // number of frames written to the + // audio HAL when this track will be fully rendered + // zero means not monitoring private: IAudioFlinger::track_flags_t mFlags; @@ -904,7 +922,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId); virtual ~TimedTrack(); @@ -947,7 +965,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId); @@ -982,11 +1000,11 @@ private: }; - // playback track + // playback track, used by DuplicatingThread class OutputTrack : public Track { public: - class Buffer: public AudioBufferProvider::Buffer { + class Buffer : public AudioBufferProvider::Buffer { public: int16_t *mBuffer; }; @@ -996,10 +1014,11 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount); + size_t frameCount); virtual ~OutputTrack(); - virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); virtual void stop(); bool write(int16_t* data, uint32_t frames); @@ -1013,7 +1032,8 @@ private: NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value }; - status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); + status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, + uint32_t waitTimeMs); void clearBufferQueue(); // Maximum number of pending buffers allocated by OutputTrack::write() @@ -1023,10 +1043,11 @@ private: AudioBufferProvider::Buffer mOutBuffer; bool mActive; DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() + void* mBuffers; // starting address of buffers in plain memory }; // end of OutputTrack - PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, - audio_io_handle_t id, audio_devices_t device, type_t type); + PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, + audio_io_handle_t id, audio_devices_t device, type_t type); virtual ~PlaybackThread(); void dump(int fd, const Vector<String16>& args); @@ -1078,10 +1099,10 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, - IAudioFlinger::track_flags_t flags, + IAudioFlinger::track_flags_t *flags, pid_t tid, status_t *status); @@ -1185,7 +1206,8 @@ public: void dumpTracks(int fd, const Vector<String16>& args); SortedVector< sp<Track> > mTracks; - // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread + // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by + // DuplicatingThread stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; AudioStreamOut *mOutput; @@ -1248,11 +1270,11 @@ public: class MixerThread : public PlaybackThread { public: - MixerThread (const sp<AudioFlinger>& audioFlinger, - AudioStreamOut* output, - audio_io_handle_t id, - audio_devices_t device, - type_t type = MIXER); + MixerThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, + audio_io_handle_t id, + audio_devices_t device, + type_t type = MIXER); virtual ~MixerThread(); // Thread virtuals @@ -1305,8 +1327,8 @@ public: class DirectOutputThread : public PlaybackThread { public: - DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, - audio_io_handle_t id, audio_devices_t device); + DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, + audio_io_handle_t id, audio_devices_t device); virtual ~DirectOutputThread(); // Thread virtuals @@ -1326,11 +1348,11 @@ public: virtual void threadLoop_mix(); virtual void threadLoop_sleepTime(); + private: // volumes last sent to audio HAL with stream->set_volume() float mLeftVolFloat; float mRightVolFloat; -private: // prepareTracks_l() tells threadLoop_mix() the name of the single active track sp<Track> mActiveTrack; public: @@ -1339,8 +1361,8 @@ private: class DuplicatingThread : public MixerThread { public: - DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, - audio_io_handle_t id); + DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, + audio_io_handle_t id); virtual ~DuplicatingThread(); // Thread virtuals @@ -1441,7 +1463,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, int sessionId); virtual ~RecordTrack(); @@ -1453,11 +1475,14 @@ private: // clear the buffer overflow flag void clearOverflow() { mOverflow = false; } // set the buffer overflow flag and return previous value - bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } + bool setOverflow() { bool tmp = mOverflow; mOverflow = true; + return tmp; } static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); + virtual bool isOut() const; + private: friend class AudioFlinger; // for mState @@ -1465,7 +1490,8 @@ private: RecordTrack& operator = (const RecordTrack&); // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer @@ -1476,7 +1502,8 @@ private: uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t id, - audio_devices_t device); + audio_devices_t device, + const sp<NBAIO_Sink>& teeSink); virtual ~RecordThread(); // no addTrack_l ? @@ -1499,7 +1526,7 @@ private: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, @@ -1572,6 +1599,9 @@ private: // when < 0, maximum frames to drop before starting capture even if sync event is // not received ssize_t mFramestoDrop; + + // For dumpsys + const sp<NBAIO_Sink> mTeeSink; }; // server side of the client's IAudioRecord @@ -1607,7 +1637,7 @@ private: // ramping when effects are activated/deactivated. // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by // the attached track(s) to accumulate their auxiliary channel. - class EffectModule: public RefBase { + class EffectModule : public RefBase { public: EffectModule(ThreadBase *thread, const wp<AudioFlinger::EffectChain>& chain, @@ -1781,7 +1811,8 @@ mutable Mutex mLock; // mutex for process, commands and handl sp<IEffectClient> mEffectClient; // callback interface for client notifications /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() sp<IMemory> mCblkMemory; // shared memory for control block - effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory + effect_param_cblk_t* mCblk; // control block for deferred parameter setting via + // shared memory uint8_t* mBuffer; // pointer to parameter area in shared memory int mPriority; // client application priority to control the effect bool mHasControl; // true if this handle is controlling the effect @@ -1794,11 +1825,11 @@ mutable Mutex mLock; // mutex for process, commands and handl // the EffectChain class represents a group of effects associated to one audio session. // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). // The EffecChain with session ID 0 contains global effects applied to the output mix. - // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) - // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding - // in the effect process order. When attached to a track (session ID != 0), it also provide it's own - // input buffer used by the track as accumulation buffer. - class EffectChain: public RefBase { + // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to + // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the + // order corresponding in the effect process order. When attached to a track (session ID != 0), + // it also provide it's own input buffer used by the track as accumulation buffer. + class EffectChain : public RefBase { public: EffectChain(const wp<ThreadBase>& wThread, int sessionId); EffectChain(ThreadBase *thread, int sessionId); @@ -2064,6 +2095,13 @@ private: // for use from destructor status_t closeOutput_nonvirtual(audio_io_handle_t output); status_t closeInput_nonvirtual(audio_io_handle_t input); + + // all record threads serially share a common tee sink, which is re-created on format change + sp<NBAIO_Sink> mRecordTeeSink; + sp<NBAIO_Source> mRecordTeeSource; + +public: + static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); }; diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index af169d5..b3ca877 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -106,8 +106,16 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u", maxNumTracks, MAX_NUM_TRACKS); + // AudioMixer is not yet capable of more than 32 active track inputs + ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS); + + // AudioMixer is not yet capable of multi-channel output beyond stereo + ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS); + LocalClock lc; + pthread_once(&sOnceControl, &sInitRoutine); + mState.enabledTracks= 0; mState.needsChanged = 0; mState.frameCount = frameCount; @@ -121,8 +129,6 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr // and mTrackNames is initially 0. However, leave it here until that's verified. track_t* t = mState.tracks; for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { - // FIXME redundant per track - t->localTimeFreq = lc.getLocalFreq(); t->resampler = NULL; t->downmixerBufferProvider = NULL; t++; @@ -192,7 +198,6 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) t->sessionId = sessionId; // setBufferProvider(name, AudioBufferProvider *) is required before enable(name) t->bufferProvider = NULL; - t->downmixerBufferProvider = NULL; t->buffer.raw = NULL; // no initialization needed // t->buffer.frameCount @@ -203,7 +208,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name) t->mainBuffer = NULL; t->auxBuffer = NULL; - // see t->localTimeFreq in constructor above + t->downmixerBufferProvider = NULL; status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask); if (status == OK) { @@ -556,7 +561,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) // the resampler sees the number of channels after the downmixer, if any downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount, devSampleRate, quality); - resampler->setLocalTimeFreq(localTimeFreq); + resampler->setLocalTimeFreq(sLocalTimeFreq); } return true; } @@ -760,7 +765,8 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) } -void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, + int32_t* temp, int32_t* aux) { t->resampler->setSampleRate(t->sampleRate); @@ -793,11 +799,13 @@ void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFram } } -void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, + int32_t* aux) { } -void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { int32_t vl = t->prevVolume[0]; int32_t vr = t->prevVolume[1]; @@ -839,7 +847,8 @@ void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, i t->adjustVolumeRamp(aux != NULL); } -void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t vl = t->volume[0]; const int16_t vr = t->volume[1]; @@ -867,7 +876,8 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32 } } -void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<const int16_t *>(t->in); @@ -957,7 +967,8 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount t->in = in; } -void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<int16_t const *>(t->in); @@ -1142,7 +1153,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) while (outFrames) { size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; if (inFrames) { - t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); + t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, + state->resampleTemp, aux); t.frameCount -= inFrames; outFrames -= inFrames; if (CC_UNLIKELY(aux != NULL)) { @@ -1151,7 +1163,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) } if (t.frameCount == 0 && outFrames) { t.bufferProvider->releaseBuffer(&t.buffer); - t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames); + t.buffer.frameCount = (state->frameCount - numFrames) - + (BLOCKSIZE - outFrames); int64_t outputPTS = calculateOutputPTS( t, pts, numFrames + (BLOCKSIZE - outFrames)); t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); @@ -1241,7 +1254,8 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) if (CC_UNLIKELY(aux != NULL)) { aux += outFrames; } - t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux); + t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, + state->resampleTemp, aux); outFrames += t.buffer.frameCount; t.bufferProvider->releaseBuffer(&t.buffer); } @@ -1281,7 +1295,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, // been enabled for mixing. if (in == NULL || ((unsigned long)in & 3)) { memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x", + ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " + "buffer %p track %d, channels %d, needs %08x", in, i, t.channelCount, t.needs); return; } @@ -1423,7 +1438,16 @@ int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, if (AudioBufferProvider::kInvalidPTS == basePTS) return AudioBufferProvider::kInvalidPTS; - return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate); + return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate); +} + +/*static*/ uint64_t AudioMixer::sLocalTimeFreq; +/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT; + +/*static*/ void AudioMixer::sInitRoutine() +{ + LocalClock lc; + sLocalTimeFreq = lc.getLocalFreq(); } // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index 6333357..fd21fda 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -41,8 +41,15 @@ public: /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed + + // This mixer has a hard-coded upper limit of 32 active track inputs. + // Adding support for > 32 tracks would require more than simply changing this value. static const uint32_t MAX_NUM_TRACKS = 32; // maximum number of channels supported by the mixer + + // This mixer has a hard-coded upper limit of 2 channels for output. + // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. + // Adding support for > 2 channel output would require more than simply changing this value. static const uint32_t MAX_NUM_CHANNELS = 2; // maximum number of channels supported for the content static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8; @@ -139,7 +146,8 @@ private: struct track_t; class DownmixerBufferProvider; - typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); + typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, + int32_t* aux); static const int BLOCKSIZE = 16; // 4 cache lines struct track_t { @@ -188,12 +196,12 @@ private: // 16-byte boundary - uint64_t localTimeFreq; - DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes int32_t sessionId; + int32_t padding[2]; + // 16-byte boundary bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); @@ -254,12 +262,17 @@ private: static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); static void unprepareTrackForDownmix(track_t* pTrack, int trackName); - static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); + static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); - static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); + static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); + static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); + static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux); + static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux); static void process__validate(state_t* state, int64_t pts); static void process__nop(state_t* state, int64_t pts); @@ -274,6 +287,10 @@ private: static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS, int outputFrameIndex); + + static uint64_t sLocalTimeFreq; + static pthread_once_t sOnceControl; + static void sInitRoutine(); }; // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 8b99bd2..ea130ba 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -227,7 +227,8 @@ audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, } ALOGV("getOutput() tid %d", gettid()); Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, flags); + return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, + flags); } status_t AudioPolicyService::startOutput(audio_io_handle_t output, @@ -280,7 +281,7 @@ audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource, Mutex::Autolock _l(mLock); // the audio_in_acoustics_t parameter is ignored by get_input() audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, - format, channelMask, (audio_in_acoustics_t) 0); + format, channelMask, (audio_in_acoustics_t) 0); if (input == 0) { return input; diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h index 63f9549..92653c1 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audioflinger/AudioPolicyService.h @@ -142,11 +142,11 @@ private: status_t dumpInternals(int fd); // Thread used for tone playback and to send audio config commands to audio flinger - // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone() - // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause - // calls to AudioPolicyService and an attempt to lock mLock. - // For audio config commands, it is necessary because audio flinger requires that the calling process (user) - // has permission to modify audio settings. + // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because + // startTone() and stopTone() are normally called with mLock locked and requesting a tone start + // or stop will cause calls to AudioPolicyService and an attempt to lock mLock. + // For audio config commands, it is necessary because audio flinger requires that the calling + // process (user) has permission to modify audio settings. class AudioCommandThread : public Thread { class AudioCommand; public: diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index ffea9b9..2c3c719 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -82,10 +82,8 @@ bool AudioResampler::qualityIsSupported(src_quality quality) switch (quality) { case DEFAULT_QUALITY: case LOW_QUALITY: -#if 0 // these have not been qualified recently so are not supported unless explicitly requested case MED_QUALITY: case HIGH_QUALITY: -#endif case VERY_HIGH_QUALITY: return true; default: @@ -190,12 +188,10 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, ALOGV("Create linear Resampler"); resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); break; -#if 0 // disabled because it has not been qualified recently, if requested will use default: case MED_QUALITY: ALOGV("Create cubic Resampler"); resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); break; -#endif case HIGH_QUALITY: ALOGV("Create HIGH_QUALITY sinc Resampler"); resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp index 9e8447a..d68b839 100644 --- a/services/audioflinger/AudioResamplerSinc.cpp +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -17,13 +17,33 @@ #define LOG_TAG "AudioResamplerSinc" //#define LOG_NDEBUG 0 +#include <malloc.h> #include <string.h> -#include "AudioResamplerSinc.h" +#include <stdlib.h> #include <dlfcn.h> + +#include <cutils/compiler.h> #include <cutils/properties.h> -#include <stdlib.h> + #include <utils/Log.h> +#include "AudioResamplerSinc.h" + + + +#if defined(__arm__) && !defined(__thumb__) +#define USE_INLINE_ASSEMBLY (true) +#else +#define USE_INLINE_ASSEMBLY (false) +#endif + +#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__) +#define USE_NEON (true) +#else +#define USE_NEON (false) +#endif + + namespace android { // ---------------------------------------------------------------------------- @@ -31,37 +51,274 @@ namespace android { /* * These coeficients are computed with the "fir" utility found in * tools/resampler_tools - * TODO: A good optimization would be to transpose this matrix, to take - * better advantage of the data-cache. + * cmd-line: fir -l 7 -s 48000 -c 20478 */ -const int32_t AudioResamplerSinc::mFirCoefsUp[] = { - 0x7fffffff, 0x7f15d078, 0x7c5e0da6, 0x77ecd867, 0x71e2e251, 0x6a6c304a, 0x61be7269, 0x58170412, 0x4db8ab05, 0x42e92ea6, 0x37eee214, 0x2d0e3bb1, 0x22879366, 0x18951e95, 0x0f693d0d, 0x072d2621, - 0x00000000, 0xf9f66655, 0xf51a5fd7, 0xf16bbd84, 0xeee0d9ac, 0xed67a922, 0xece70de6, 0xed405897, 0xee50e505, 0xeff3be30, 0xf203370f, 0xf45a6741, 0xf6d67d53, 0xf957db66, 0xfbc2f647, 0xfe00f2b9, - 0x00000000, 0x01b37218, 0x0313a0c6, 0x041d930d, 0x04d28057, 0x053731b0, 0x05534dff, 0x05309bfd, 0x04da440d, 0x045c1aee, 0x03c1fcdd, 0x03173ef5, 0x02663ae8, 0x01b7f736, 0x0113ec79, 0x007fe6a9, - 0x00000000, 0xff96b229, 0xff44f99f, 0xff0a86be, 0xfee5f803, 0xfed518fd, 0xfed521fd, 0xfee2f4fd, 0xfefb54f8, 0xff1b159b, 0xff3f4203, 0xff6539e0, 0xff8ac502, 0xffae1ddd, 0xffcdf3f9, 0xffe96798, - 0x00000000, 0x00119de6, 0x001e6b7e, 0x0026cb7a, 0x002b4830, 0x002c83d6, 0x002b2a82, 0x0027e67a, 0x002356f9, 0x001e098e, 0x001875e4, 0x0012fbbe, 0x000de2d1, 0x00095c10, 0x00058414, 0x00026636, - 0x00000000, 0xfffe44a9, 0xfffd206d, 0xfffc7b7f, 0xfffc3c8f, 0xfffc4ac2, 0xfffc8f2b, 0xfffcf5c4, 0xfffd6df3, 0xfffdeab2, 0xfffe6275, 0xfffececf, 0xffff2c07, 0xffff788c, 0xffffb471, 0xffffe0f2, - 0x00000000, 0x000013e6, 0x00001f03, 0x00002396, 0x00002399, 0x000020b6, 0x00001c3c, 0x00001722, 0x00001216, 0x00000d81, 0x0000099c, 0x0000067c, 0x00000419, 0x0000025f, 0x00000131, 0x00000070, - 0x00000000, 0xffffffc7, 0xffffffb3, 0xffffffb3, 0xffffffbe, 0xffffffcd, 0xffffffdb, 0xffffffe7, 0xfffffff0, 0xfffffff7, 0xfffffffb, 0xfffffffe, 0xffffffff, 0x00000000, 0x00000000, 0x00000000, - 0x00000000 // this one is needed for lerping the last coefficient +const int32_t AudioResamplerSinc::mFirCoefsUp[] __attribute__ ((aligned (32))) = { + 0x6d374bc7, 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300, + 0x6d35278a, 0x103e8192, 0xf36b9dfd, 0x07bdfaa5, 0xfc5102d0, 0x013d618d, 0xffc663b9, 0xfffd9592, + 0x6d2ebafe, 0x0f62811a, 0xf3b3d8ac, 0x07a9f399, 0xfc51d9a6, 0x0140bea5, 0xffc41212, 0xfffe631e, + 0x6d24069d, 0x0e8875ad, 0xf3fcb43e, 0x07953976, 0xfc53216f, 0x0143e67c, 0xffc1d373, 0xffff2b9f, + 0x6d150b35, 0x0db06a89, 0xf4462690, 0x077fd0ac, 0xfc54d8ae, 0x0146d965, 0xffbfa7d9, 0xffffef10, + 0x6d01c9e3, 0x0cda6ab5, 0xf4902587, 0x0769bdaf, 0xfc56fdda, 0x014997bb, 0xffbd8f40, 0x0000ad6e, + 0x6cea4418, 0x0c0680fe, 0xf4daa718, 0x07530501, 0xfc598f60, 0x014c21db, 0xffbb89a1, 0x000166b6, + 0x6cce7b97, 0x0b34b7f5, 0xf525a143, 0x073bab28, 0xfc5c8ba5, 0x014e782a, 0xffb996f3, 0x00021ae5, + 0x6cae7272, 0x0a6519f4, 0xf5710a17, 0x0723b4b4, 0xfc5ff105, 0x01509b14, 0xffb7b728, 0x0002c9fd, + 0x6c8a2b0f, 0x0997b116, 0xf5bcd7b1, 0x070b2639, 0xfc63bdd3, 0x01528b08, 0xffb5ea31, 0x000373fb, + 0x6c61a823, 0x08cc873c, 0xf609003f, 0x06f20453, 0xfc67f05a, 0x0154487b, 0xffb42ffc, 0x000418e2, + 0x6c34ecb5, 0x0803a60a, 0xf6557a00, 0x06d853a2, 0xfc6c86dd, 0x0155d3e8, 0xffb28876, 0x0004b8b3, + 0x6c03fc1c, 0x073d16e7, 0xf6a23b44, 0x06be18cd, 0xfc717f97, 0x01572dcf, 0xffb0f388, 0x00055371, + 0x6bced9ff, 0x0678e2fc, 0xf6ef3a6e, 0x06a3587e, 0xfc76d8bc, 0x015856b6, 0xffaf7118, 0x0005e921, + 0x6b958a54, 0x05b71332, 0xf73c6df4, 0x06881761, 0xfc7c9079, 0x01594f25, 0xffae010b, 0x000679c5, + 0x6b581163, 0x04f7b037, 0xf789cc61, 0x066c5a27, 0xfc82a4f4, 0x015a17ab, 0xffaca344, 0x00070564, + 0x6b1673c1, 0x043ac276, 0xf7d74c53, 0x06502583, 0xfc89144d, 0x015ab0db, 0xffab57a1, 0x00078c04, + 0x6ad0b652, 0x0380521c, 0xf824e480, 0x06337e2a, 0xfc8fdc9f, 0x015b1b4e, 0xffaa1e02, 0x00080dab, + 0x6a86de48, 0x02c86715, 0xf8728bb3, 0x061668d2, 0xfc96fbfc, 0x015b579e, 0xffa8f641, 0x00088a62, + 0x6a38f123, 0x0213090c, 0xf8c038d0, 0x05f8ea30, 0xfc9e7074, 0x015b666c, 0xffa7e039, 0x00090230, + 0x69e6f4b1, 0x01603f6e, 0xf90de2d1, 0x05db06fc, 0xfca63810, 0x015b485b, 0xffa6dbc0, 0x0009751e, + 0x6990ef0b, 0x00b01162, 0xf95b80cb, 0x05bcc3ed, 0xfcae50d6, 0x015afe14, 0xffa5e8ad, 0x0009e337, + 0x6936e697, 0x000285d0, 0xf9a909ea, 0x059e25b5, 0xfcb6b8c4, 0x015a8843, 0xffa506d2, 0x000a4c85, + 0x68d8e206, 0xff57a35e, 0xf9f67577, 0x057f310a, 0xfcbf6dd8, 0x0159e796, 0xffa43603, 0x000ab112, + 0x6876e855, 0xfeaf706f, 0xfa43bad2, 0x055fea9d, 0xfcc86e09, 0x01591cc0, 0xffa3760e, 0x000b10ec, + 0x681100c9, 0xfe09f323, 0xfa90d17b, 0x0540571a, 0xfcd1b74c, 0x01582878, 0xffa2c6c2, 0x000b6c1d, + 0x67a732f4, 0xfd673159, 0xfaddb10c, 0x05207b2f, 0xfcdb4793, 0x01570b77, 0xffa227ec, 0x000bc2b3, + 0x673986ac, 0xfcc730aa, 0xfb2a513b, 0x05005b82, 0xfce51ccb, 0x0155c678, 0xffa19957, 0x000c14bb, + 0x66c80413, 0xfc29f670, 0xfb76a9dd, 0x04dffcb6, 0xfcef34e1, 0x01545a3c, 0xffa11acb, 0x000c6244, + 0x6652b392, 0xfb8f87bd, 0xfbc2b2e4, 0x04bf6369, 0xfcf98dbe, 0x0152c783, 0xffa0ac11, 0x000cab5c, + 0x65d99dd5, 0xfaf7e963, 0xfc0e6461, 0x049e9433, 0xfd04254a, 0x01510f13, 0xffa04cf0, 0x000cf012, + 0x655ccbd3, 0xfa631fef, 0xfc59b685, 0x047d93a8, 0xfd0ef969, 0x014f31b2, 0xff9ffd2c, 0x000d3075, + 0x64dc46c3, 0xf9d12fab, 0xfca4a19f, 0x045c6654, 0xfd1a0801, 0x014d3029, 0xff9fbc89, 0x000d6c97, + 0x64581823, 0xf9421c9d, 0xfcef1e20, 0x043b10bd, 0xfd254ef4, 0x014b0b45, 0xff9f8ac9, 0x000da486, + 0x63d049b4, 0xf8b5ea87, 0xfd392498, 0x04199760, 0xfd30cc24, 0x0148c3d2, 0xff9f67ae, 0x000dd854, + 0x6344e578, 0xf82c9ce7, 0xfd82adba, 0x03f7feb4, 0xfd3c7d73, 0x01465a9f, 0xff9f52f7, 0x000e0812, + 0x62b5f5b2, 0xf7a636fa, 0xfdcbb25a, 0x03d64b27, 0xfd4860c2, 0x0143d07f, 0xff9f4c65, 0x000e33d3, + 0x622384e8, 0xf722bbb5, 0xfe142b6e, 0x03b4811d, 0xfd5473f3, 0x01412643, 0xff9f53b4, 0x000e5ba7, + 0x618d9ddc, 0xf6a22dcf, 0xfe5c120f, 0x0392a4f4, 0xfd60b4e7, 0x013e5cc0, 0xff9f68a1, 0x000e7fa1, + 0x60f44b91, 0xf6248fb6, 0xfea35f79, 0x0370bafc, 0xfd6d2180, 0x013b74ca, 0xff9f8ae9, 0x000e9fd5, + 0x60579947, 0xf5a9e398, 0xfeea0d0c, 0x034ec77f, 0xfd79b7a1, 0x01386f3a, 0xff9fba47, 0x000ebc54, + 0x5fb79278, 0xf5322b61, 0xff30144a, 0x032ccebb, 0xfd86752e, 0x01354ce7, 0xff9ff674, 0x000ed533, + 0x5f1442dc, 0xf4bd68b6, 0xff756edc, 0x030ad4e1, 0xfd93580d, 0x01320ea9, 0xffa03f2b, 0x000eea84, + 0x5e6db665, 0xf44b9cfe, 0xffba168d, 0x02e8de19, 0xfda05e23, 0x012eb55a, 0xffa09425, 0x000efc5c, + 0x5dc3f93c, 0xf3dcc959, 0xfffe054e, 0x02c6ee7f, 0xfdad855b, 0x012b41d3, 0xffa0f519, 0x000f0ace, + 0x5d1717c4, 0xf370eea9, 0x00413536, 0x02a50a22, 0xfdbacb9e, 0x0127b4f1, 0xffa161bf, 0x000f15ef, + 0x5c671e96, 0xf3080d8c, 0x0083a081, 0x02833506, 0xfdc82edb, 0x01240f8e, 0xffa1d9cf, 0x000f1dd2, + 0x5bb41a80, 0xf2a2265e, 0x00c54190, 0x02617321, 0xfdd5ad01, 0x01205285, 0xffa25cfe, 0x000f228d, + 0x5afe1886, 0xf23f393b, 0x010612eb, 0x023fc85c, 0xfde34403, 0x011c7eb2, 0xffa2eb04, 0x000f2434, + 0x5a4525df, 0xf1df45fd, 0x01460f41, 0x021e3891, 0xfdf0f1d6, 0x011894f0, 0xffa38395, 0x000f22dc, + 0x59894ff3, 0xf1824c3e, 0x01853165, 0x01fcc78f, 0xfdfeb475, 0x0114961b, 0xffa42668, 0x000f1e99, + 0x58caa45b, 0xf1284b58, 0x01c37452, 0x01db7914, 0xfe0c89db, 0x0110830f, 0xffa4d332, 0x000f1781, + 0x580930e1, 0xf0d14267, 0x0200d32c, 0x01ba50d2, 0xfe1a7009, 0x010c5ca6, 0xffa589a6, 0x000f0da8, + 0x5745037c, 0xf07d3043, 0x023d493c, 0x0199526b, 0xfe286505, 0x010823ba, 0xffa6497c, 0x000f0125, + 0x567e2a51, 0xf02c138a, 0x0278d1f2, 0x01788170, 0xfe3666d5, 0x0103d927, 0xffa71266, 0x000ef20b, + 0x55b4b3af, 0xefddea9a, 0x02b368e6, 0x0157e166, 0xfe447389, 0x00ff7dc4, 0xffa7e41a, 0x000ee070, + 0x54e8ae13, 0xef92b393, 0x02ed09d7, 0x013775bf, 0xfe528931, 0x00fb126b, 0xffa8be4c, 0x000ecc69, + 0x541a281e, 0xef4a6c58, 0x0325b0ad, 0x011741df, 0xfe60a5e5, 0x00f697f3, 0xffa9a0b1, 0x000eb60b, + 0x5349309e, 0xef051290, 0x035d5977, 0x00f7491a, 0xfe6ec7c0, 0x00f20f32, 0xffaa8afe, 0x000e9d6b, + 0x5275d684, 0xeec2a3a3, 0x0394006a, 0x00d78eb3, 0xfe7cece2, 0x00ed78ff, 0xffab7ce7, 0x000e829e, + 0x51a028e8, 0xee831cc3, 0x03c9a1e5, 0x00b815da, 0xfe8b1373, 0x00e8d62d, 0xffac7621, 0x000e65ba, + 0x50c83704, 0xee467ae1, 0x03fe3a6f, 0x0098e1b3, 0xfe99399f, 0x00e4278f, 0xffad7662, 0x000e46d3, + 0x4fee1037, 0xee0cbab9, 0x0431c6b5, 0x0079f54c, 0xfea75d97, 0x00df6df7, 0xffae7d5f, 0x000e25fd, + 0x4f11c3fe, 0xedd5d8ca, 0x0464438c, 0x005b53a4, 0xfeb57d92, 0x00daaa34, 0xffaf8acd, 0x000e034f, + 0x4e3361f7, 0xeda1d15c, 0x0495adf2, 0x003cffa9, 0xfec397cf, 0x00d5dd16, 0xffb09e63, 0x000ddedb, + 0x4d52f9df, 0xed70a07d, 0x04c6030d, 0x001efc35, 0xfed1aa92, 0x00d10769, 0xffb1b7d8, 0x000db8b7, + 0x4c709b8e, 0xed424205, 0x04f54029, 0x00014c12, 0xfedfb425, 0x00cc29f7, 0xffb2d6e1, 0x000d90f6, + 0x4b8c56f8, 0xed16b196, 0x052362ba, 0xffe3f1f7, 0xfeedb2da, 0x00c7458a, 0xffb3fb37, 0x000d67ae, + 0x4aa63c2c, 0xecedea99, 0x0550685d, 0xffc6f08a, 0xfefba508, 0x00c25ae8, 0xffb52490, 0x000d3cf1, + 0x49be5b50, 0xecc7e845, 0x057c4ed4, 0xffaa4a5d, 0xff09890f, 0x00bd6ad7, 0xffb652a7, 0x000d10d5, + 0x48d4c4a2, 0xeca4a59b, 0x05a7140b, 0xff8e01f1, 0xff175d53, 0x00b87619, 0xffb78533, 0x000ce36b, + 0x47e98874, 0xec841d68, 0x05d0b612, 0xff7219b3, 0xff252042, 0x00b37d70, 0xffb8bbed, 0x000cb4c8, + 0x46fcb72d, 0xec664a48, 0x05f93324, 0xff5693fe, 0xff32d04f, 0x00ae8198, 0xffb9f691, 0x000c84ff, + 0x460e6148, 0xec4b26a2, 0x0620899e, 0xff3b731b, 0xff406bf8, 0x00a9834e, 0xffbb34d8, 0x000c5422, + 0x451e9750, 0xec32acb0, 0x0646b808, 0xff20b93e, 0xff4df1be, 0x00a4834c, 0xffbc767f, 0x000c2245, + 0x442d69de, 0xec1cd677, 0x066bbd0d, 0xff066889, 0xff5b602c, 0x009f8249, 0xffbdbb42, 0x000bef79, + 0x433ae99c, 0xec099dcf, 0x068f9781, 0xfeec830d, 0xff68b5d5, 0x009a80f8, 0xffbf02dd, 0x000bbbd2, + 0x4247273f, 0xebf8fc64, 0x06b2465b, 0xfed30ac5, 0xff75f153, 0x0095800c, 0xffc04d0f, 0x000b8760, + 0x41523389, 0xebeaebaf, 0x06d3c8bb, 0xfeba0199, 0xff831148, 0x00908034, 0xffc19996, 0x000b5235, + 0x405c1f43, 0xebdf6500, 0x06f41de3, 0xfea16960, 0xff90145e, 0x008b821b, 0xffc2e832, 0x000b1c64, + 0x3f64fb40, 0xebd6617b, 0x0713453d, 0xfe8943dc, 0xff9cf947, 0x0086866b, 0xffc438a3, 0x000ae5fc, + 0x3e6cd85b, 0xebcfda19, 0x07313e56, 0xfe7192bd, 0xffa9bebe, 0x00818dcb, 0xffc58aaa, 0x000aaf0f, + 0x3d73c772, 0xebcbc7a7, 0x074e08e0, 0xfe5a579d, 0xffb66386, 0x007c98de, 0xffc6de09, 0x000a77ac, + 0x3c79d968, 0xebca22cc, 0x0769a4b2, 0xfe439407, 0xffc2e669, 0x0077a845, 0xffc83285, 0x000a3fe5, + 0x3b7f1f23, 0xebcae405, 0x078411c7, 0xfe2d496f, 0xffcf463a, 0x0072bc9d, 0xffc987e0, 0x000a07c9, + 0x3a83a989, 0xebce03aa, 0x079d503b, 0xfe177937, 0xffdb81d6, 0x006dd680, 0xffcadde1, 0x0009cf67, + 0x3987897f, 0xebd379eb, 0x07b56051, 0xfe0224b0, 0xffe79820, 0x0068f687, 0xffcc344c, 0x000996ce, + 0x388acfe9, 0xebdb3ed5, 0x07cc426c, 0xfded4d13, 0xfff38806, 0x00641d44, 0xffcd8aeb, 0x00095e0e, + 0x378d8da8, 0xebe54a4f, 0x07e1f712, 0xfdd8f38b, 0xffff507b, 0x005f4b4a, 0xffcee183, 0x00092535, + 0x368fd397, 0xebf1941f, 0x07f67eec, 0xfdc5192d, 0x000af07f, 0x005a8125, 0xffd037e0, 0x0008ec50, + 0x3591b28b, 0xec0013e8, 0x0809dac3, 0xfdb1befc, 0x00166718, 0x0055bf60, 0xffd18dcc, 0x0008b36e, + 0x34933b50, 0xec10c12c, 0x081c0b84, 0xfd9ee5e7, 0x0021b355, 0x00510682, 0xffd2e311, 0x00087a9c, + 0x33947eab, 0xec23934f, 0x082d1239, 0xfd8c8ecc, 0x002cd44d, 0x004c570f, 0xffd4377d, 0x000841e8, + 0x32958d55, 0xec388194, 0x083cf010, 0xfd7aba74, 0x0037c922, 0x0047b186, 0xffd58ade, 0x0008095d, + 0x319677fa, 0xec4f8322, 0x084ba654, 0xfd696998, 0x004290fc, 0x00431666, 0xffd6dd02, 0x0007d108, + 0x30974f3b, 0xec688f02, 0x08593671, 0xfd589cdc, 0x004d2b0e, 0x003e8628, 0xffd82dba, 0x000798f5, + 0x2f9823a8, 0xec839c22, 0x0865a1f1, 0xfd4854d3, 0x00579691, 0x003a0141, 0xffd97cd6, 0x00076130, + 0x2e9905c1, 0xeca0a156, 0x0870ea7e, 0xfd3891fd, 0x0061d2ca, 0x00358824, 0xffdaca2a, 0x000729c4, + 0x2d9a05f4, 0xecbf9558, 0x087b11de, 0xfd2954c8, 0x006bdf05, 0x00311b41, 0xffdc1588, 0x0006f2bb, + 0x2c9b349e, 0xece06ecb, 0x088419f6, 0xfd1a9d91, 0x0075ba95, 0x002cbb03, 0xffdd5ec6, 0x0006bc21, + 0x2b9ca203, 0xed032439, 0x088c04c8, 0xfd0c6ca2, 0x007f64da, 0x002867d2, 0xffdea5bb, 0x000685ff, + 0x2a9e5e57, 0xed27ac16, 0x0892d470, 0xfcfec233, 0x0088dd38, 0x00242213, 0xffdfea3c, 0x0006505f, + 0x29a079b2, 0xed4dfcc2, 0x08988b2a, 0xfcf19e6b, 0x0092231e, 0x001fea27, 0xffe12c22, 0x00061b4b, + 0x28a30416, 0xed760c88, 0x089d2b4a, 0xfce50161, 0x009b3605, 0x001bc06b, 0xffe26b48, 0x0005e6cb, + 0x27a60d6a, 0xed9fd1a2, 0x08a0b740, 0xfcd8eb17, 0x00a4156b, 0x0017a53b, 0xffe3a788, 0x0005b2e8, + 0x26a9a57b, 0xedcb4237, 0x08a33196, 0xfccd5b82, 0x00acc0da, 0x001398ec, 0xffe4e0bf, 0x00057faa, + 0x25addbf9, 0xedf8545b, 0x08a49cf0, 0xfcc25285, 0x00b537e1, 0x000f9bd2, 0xffe616c8, 0x00054d1a, + 0x24b2c075, 0xee26fe17, 0x08a4fc0d, 0xfcb7cff0, 0x00bd7a1c, 0x000bae3c, 0xffe74984, 0x00051b3e, + 0x23b86263, 0xee573562, 0x08a451c0, 0xfcadd386, 0x00c5872a, 0x0007d075, 0xffe878d3, 0x0004ea1d, + 0x22bed116, 0xee88f026, 0x08a2a0f8, 0xfca45cf7, 0x00cd5eb7, 0x000402c8, 0xffe9a494, 0x0004b9c0, + 0x21c61bc0, 0xeebc2444, 0x089fecbb, 0xfc9b6be5, 0x00d50075, 0x00004579, 0xffeaccaa, 0x00048a2b, + 0x20ce516f, 0xeef0c78d, 0x089c3824, 0xfc92ffe1, 0x00dc6c1e, 0xfffc98c9, 0xffebf0fa, 0x00045b65, + 0x1fd7810f, 0xef26cfca, 0x08978666, 0xfc8b186d, 0x00e3a175, 0xfff8fcf7, 0xffed1166, 0x00042d74, + 0x1ee1b965, 0xef5e32bd, 0x0891dac8, 0xfc83b4fc, 0x00eaa045, 0xfff5723d, 0xffee2dd7, 0x0004005e, + 0x1ded0911, 0xef96e61c, 0x088b38a9, 0xfc7cd4f0, 0x00f16861, 0xfff1f8d2, 0xffef4632, 0x0003d426, + 0x1cf97e8b, 0xefd0df9a, 0x0883a378, 0xfc76779e, 0x00f7f9a3, 0xffee90eb, 0xfff05a60, 0x0003a8d2, + 0x1c072823, 0xf00c14e1, 0x087b1ebc, 0xfc709c4d, 0x00fe53ef, 0xffeb3ab8, 0xfff16a4a, 0x00037e65, + 0x1b1613ff, 0xf0487b98, 0x0871ae0d, 0xfc6b4233, 0x0104772e, 0xffe7f666, 0xfff275db, 0x000354e5, + 0x1a26501b, 0xf0860962, 0x08675516, 0xfc66687a, 0x010a6353, 0xffe4c41e, 0xfff37d00, 0x00032c54, + 0x1937ea47, 0xf0c4b3e0, 0x085c1794, 0xfc620e3d, 0x01101858, 0xffe1a408, 0xfff47fa5, 0x000304b7, + 0x184af025, 0xf10470b0, 0x084ff957, 0xfc5e328c, 0x0115963d, 0xffde9646, 0xfff57db8, 0x0002de0e, + 0x175f6f2b, 0xf1453571, 0x0842fe3d, 0xfc5ad465, 0x011add0b, 0xffdb9af8, 0xfff67729, 0x0002b85f, + 0x1675749e, 0xf186f7c0, 0x08352a35, 0xfc57f2be, 0x011fecd3, 0xffd8b23b, 0xfff76be9, 0x000293aa, + 0x158d0d95, 0xf1c9ad40, 0x0826813e, 0xfc558c7c, 0x0124c5ab, 0xffd5dc28, 0xfff85be8, 0x00026ff2, + 0x14a646f6, 0xf20d4b92, 0x08170767, 0xfc53a07b, 0x012967b1, 0xffd318d6, 0xfff9471b, 0x00024d39, + 0x13c12d73, 0xf251c85d, 0x0806c0cb, 0xfc522d88, 0x012dd30a, 0xffd06858, 0xfffa2d74, 0x00022b7f, + 0x12ddcd8f, 0xf297194d, 0x07f5b193, 0xfc513266, 0x013207e4, 0xffcdcabe, 0xfffb0ee9, 0x00020ac7, + 0x11fc3395, 0xf2dd3411, 0x07e3ddf7, 0xfc50adcc, 0x01360670, 0xffcb4014, 0xfffbeb70, 0x0001eb10, + 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300, 0x0001cc5c, }; /* - * These coefficients are optimized for 48KHz -> 44.1KHz (stop-band at 22.050KHz) - * It's possible to use the above coefficient for any down-sampling - * at the expense of a slower processing loop (we can interpolate - * these coefficient from the above by "Stretching" them in time). + * These coefficients are optimized for 48KHz -> 44.1KHz + * cmd-line: fir -l 7 -s 48000 -c 17189 */ -const int32_t AudioResamplerSinc::mFirCoefsDown[] = { - 0x7fffffff, 0x7f55e46d, 0x7d5b4c60, 0x7a1b4b98, 0x75a7fb14, 0x7019f0bd, 0x698f875a, 0x622bfd59, 0x5a167256, 0x5178cc54, 0x487e8e6c, 0x3f53aae8, 0x36235ad4, 0x2d17047b, 0x245539ab, 0x1c00d540, - 0x14383e57, 0x0d14d5ca, 0x06aa910b, 0x0107c38b, 0xfc351654, 0xf835abae, 0xf5076b45, 0xf2a37202, 0xf0fe9faa, 0xf00a3bbd, 0xefb4aa81, 0xefea2b05, 0xf0959716, 0xf1a11e83, 0xf2f6f7a0, 0xf481fff4, - 0xf62e48ce, 0xf7e98ca5, 0xf9a38b4c, 0xfb4e4bfa, 0xfcde456f, 0xfe4a6d30, 0xff8c2fdf, 0x009f5555, 0x0181d393, 0x0233940f, 0x02b62f06, 0x030ca07d, 0x033afa62, 0x03461725, 0x03334f83, 0x030835fa, - 0x02ca59cc, 0x027f12d1, 0x022b570d, 0x01d39a49, 0x017bb78f, 0x0126e414, 0x00d7aaaf, 0x008feec7, 0x0050f584, 0x001b73e3, 0xffefa063, 0xffcd46ed, 0xffb3ddcd, 0xffa29aaa, 0xff988691, 0xff949066, - 0xff959d24, 0xff9a959e, 0xffa27195, 0xffac4011, 0xffb72d2b, 0xffc28569, 0xffcdb706, 0xffd85171, 0xffe20364, 0xffea97e9, 0xfff1f2b2, 0xfff80c06, 0xfffcec92, 0x0000a955, 0x00035fd8, 0x000532cf, - 0x00064735, 0x0006c1f9, 0x0006c62d, 0x000673ba, 0x0005e68f, 0x00053630, 0x000475a3, 0x0003b397, 0x0002fac1, 0x00025257, 0x0001be9e, 0x0001417a, 0x0000dafd, 0x000089eb, 0x00004c28, 0x00001f1d, - 0x00000000, 0xffffec10, 0xffffe0be, 0xffffdbc5, 0xffffdb39, 0xffffdd8b, 0xffffe182, 0xffffe638, 0xffffeb0a, 0xffffef8f, 0xfffff38b, 0xfffff6e3, 0xfffff993, 0xfffffba6, 0xfffffd30, 0xfffffe4a, - 0xffffff09, 0xffffff85, 0xffffffd1, 0xfffffffb, 0x0000000f, 0x00000016, 0x00000015, 0x00000012, 0x0000000d, 0x00000009, 0x00000006, 0x00000003, 0x00000002, 0x00000001, 0x00000000, 0x00000000, - 0x00000000 // this one is needed for lerping the last coefficient +const int32_t AudioResamplerSinc::mFirCoefsDown[] __attribute__ ((aligned (32))) = { + 0x5bacb6f4, 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631, + 0x5bab6c81, 0x1d3ddccd, 0xf0421d2c, 0x03af9995, 0x01818dc9, 0xfe6bb63e, 0x0079812a, 0xfffdc37d, + 0x5ba78d37, 0x1c8f2cf9, 0xf04beb1d, 0x03c9a04a, 0x016f8aca, 0xfe70a511, 0x0079e34d, 0xfffd2545, + 0x5ba1194f, 0x1be11231, 0xf056f2c7, 0x03e309fe, 0x015d9e64, 0xfe75a79f, 0x007a36e2, 0xfffc8b86, + 0x5b981122, 0x1b3393f8, 0xf0632fb7, 0x03fbd625, 0x014bc9fa, 0xfe7abd23, 0x007a7c20, 0xfffbf639, + 0x5b8c7530, 0x1a86b9bf, 0xf0709d74, 0x04140449, 0x013a0ee9, 0xfe7fe4db, 0x007ab33d, 0xfffb655b, + 0x5b7e461a, 0x19da8ae5, 0xf07f3776, 0x042b93fd, 0x01286e86, 0xfe851e05, 0x007adc72, 0xfffad8e4, + 0x5b6d84a8, 0x192f0eb7, 0xf08ef92d, 0x044284e6, 0x0116ea22, 0xfe8a67dd, 0x007af7f6, 0xfffa50ce, + 0x5b5a31c6, 0x18844c70, 0xf09fddfe, 0x0458d6b7, 0x01058306, 0xfe8fc1a5, 0x007b0603, 0xfff9cd12, + 0x5b444e81, 0x17da4b37, 0xf0b1e143, 0x046e8933, 0x00f43a74, 0xfe952a9b, 0x007b06d4, 0xfff94da9, + 0x5b2bdc0e, 0x17311222, 0xf0c4fe50, 0x04839c29, 0x00e311a9, 0xfe9aa201, 0x007afaa1, 0xfff8d28c, + 0x5b10dbc2, 0x1688a832, 0xf0d9306d, 0x04980f79, 0x00d209db, 0xfea02719, 0x007ae1a7, 0xfff85bb1, + 0x5af34f18, 0x15e11453, 0xf0ee72db, 0x04abe310, 0x00c12439, 0xfea5b926, 0x007abc20, 0xfff7e910, + 0x5ad337af, 0x153a5d5e, 0xf104c0d2, 0x04bf16e9, 0x00b061eb, 0xfeab576d, 0x007a8a49, 0xfff77a9f, + 0x5ab09748, 0x14948a16, 0xf11c1583, 0x04d1ab0d, 0x009fc413, 0xfeb10134, 0x007a4c5d, 0xfff71057, + 0x5a8b6fc7, 0x13efa12c, 0xf1346c17, 0x04e39f93, 0x008f4bcb, 0xfeb6b5c0, 0x007a029a, 0xfff6aa2b, + 0x5a63c336, 0x134ba937, 0xf14dbfb1, 0x04f4f4a2, 0x007efa29, 0xfebc745c, 0x0079ad3d, 0xfff64812, + 0x5a3993c0, 0x12a8a8bb, 0xf1680b6e, 0x0505aa6a, 0x006ed038, 0xfec23c50, 0x00794c82, 0xfff5ea02, + 0x5a0ce3b2, 0x1206a625, 0xf1834a63, 0x0515c12d, 0x005ecf01, 0xfec80ce8, 0x0078e0a9, 0xfff58ff0, + 0x59ddb57f, 0x1165a7cc, 0xf19f77a0, 0x05253938, 0x004ef782, 0xfecde571, 0x007869ee, 0xfff539cf, + 0x59ac0bba, 0x10c5b3ef, 0xf1bc8e31, 0x053412e4, 0x003f4ab4, 0xfed3c538, 0x0077e891, 0xfff4e794, + 0x5977e919, 0x1026d0b8, 0xf1da891b, 0x05424e9b, 0x002fc98a, 0xfed9ab8f, 0x00775ccf, 0xfff49934, + 0x59415075, 0x0f890437, 0xf1f96360, 0x054feccf, 0x002074ed, 0xfedf97c6, 0x0076c6e8, 0xfff44ea3, + 0x590844c9, 0x0eec5465, 0xf21917ff, 0x055cee03, 0x00114dc3, 0xfee58932, 0x00762719, 0xfff407d2, + 0x58ccc930, 0x0e50c723, 0xf239a1ef, 0x056952c3, 0x000254e8, 0xfeeb7f27, 0x00757da3, 0xfff3c4b7, + 0x588ee0ea, 0x0db6623b, 0xf25afc29, 0x05751baa, 0xfff38b32, 0xfef178fc, 0x0074cac4, 0xfff38542, + 0x584e8f56, 0x0d1d2b5d, 0xf27d219f, 0x0580495c, 0xffe4f171, 0xfef7760c, 0x00740ebb, 0xfff34968, + 0x580bd7f4, 0x0c85281f, 0xf2a00d43, 0x058adc8d, 0xffd6886d, 0xfefd75af, 0x007349c7, 0xfff3111b, + 0x57c6be67, 0x0bee5dff, 0xf2c3ba04, 0x0594d5fa, 0xffc850e6, 0xff037744, 0x00727c27, 0xfff2dc4c, + 0x577f4670, 0x0b58d262, 0xf2e822ce, 0x059e366c, 0xffba4b98, 0xff097a29, 0x0071a61b, 0xfff2aaef, + 0x573573f2, 0x0ac48a92, 0xf30d428e, 0x05a6feb9, 0xffac7936, 0xff0f7dbf, 0x0070c7e1, 0xfff27cf3, + 0x56e94af1, 0x0a318bc1, 0xf333142f, 0x05af2fbf, 0xff9eda6d, 0xff15816a, 0x006fe1b8, 0xfff2524c, + 0x569acf90, 0x099fdb04, 0xf359929a, 0x05b6ca6b, 0xff916fe1, 0xff1b848e, 0x006ef3df, 0xfff22aea, + 0x564a0610, 0x090f7d57, 0xf380b8ba, 0x05bdcfb2, 0xff843a32, 0xff218692, 0x006dfe94, 0xfff206bf, + 0x55f6f2d3, 0x0880779d, 0xf3a88179, 0x05c44095, 0xff7739f7, 0xff2786e1, 0x006d0217, 0xfff1e5bb, + 0x55a19a5c, 0x07f2ce9b, 0xf3d0e7c2, 0x05ca1e1f, 0xff6a6fc1, 0xff2d84e5, 0x006bfea4, 0xfff1c7d0, + 0x554a0148, 0x076686fc, 0xf3f9e680, 0x05cf6965, 0xff5ddc1a, 0xff33800e, 0x006af47b, 0xfff1acef, + 0x54f02c56, 0x06dba551, 0xf42378a0, 0x05d42387, 0xff517f86, 0xff3977cb, 0x0069e3d9, 0xfff19508, + 0x54942061, 0x06522e0f, 0xf44d9912, 0x05d84daf, 0xff455a80, 0xff3f6b8f, 0x0068ccfa, 0xfff1800b, + 0x5435e263, 0x05ca258f, 0xf47842c5, 0x05dbe90f, 0xff396d7f, 0xff455acf, 0x0067b01e, 0xfff16de9, + 0x53d57774, 0x0543900d, 0xf4a370ad, 0x05def6e4, 0xff2db8f2, 0xff4b4503, 0x00668d80, 0xfff15e93, + 0x5372e4c6, 0x04be71ab, 0xf4cf1dbf, 0x05e17873, 0xff223d40, 0xff5129a3, 0x0065655d, 0xfff151f9, + 0x530e2fac, 0x043ace6e, 0xf4fb44f4, 0x05e36f0d, 0xff16faca, 0xff57082e, 0x006437f1, 0xfff1480b, + 0x52a75d90, 0x03b8aa40, 0xf527e149, 0x05e4dc08, 0xff0bf1ed, 0xff5ce021, 0x00630577, 0xfff140b9, + 0x523e73fd, 0x033808eb, 0xf554edbd, 0x05e5c0c6, 0xff0122fc, 0xff62b0fd, 0x0061ce2c, 0xfff13bf3, + 0x51d37897, 0x02b8ee22, 0xf5826555, 0x05e61eae, 0xfef68e45, 0xff687a47, 0x00609249, 0xfff139aa, + 0x5166711c, 0x023b5d76, 0xf5b0431a, 0x05e5f733, 0xfeec340f, 0xff6e3b84, 0x005f520a, 0xfff139cd, + 0x50f76368, 0x01bf5a5e, 0xf5de8218, 0x05e54bcd, 0xfee2149b, 0xff73f43d, 0x005e0da8, 0xfff13c4c, + 0x5086556f, 0x0144e834, 0xf60d1d63, 0x05e41dfe, 0xfed83023, 0xff79a3fe, 0x005cc55c, 0xfff14119, + 0x50134d3e, 0x00cc0a36, 0xf63c1012, 0x05e26f4e, 0xfece86db, 0xff7f4a54, 0x005b7961, 0xfff14821, + 0x4f9e50ff, 0x0054c382, 0xf66b5544, 0x05e0414d, 0xfec518f1, 0xff84e6d0, 0x005a29ed, 0xfff15156, + 0x4f2766f2, 0xffdf171b, 0xf69ae81d, 0x05dd9593, 0xfebbe68c, 0xff8a7905, 0x0058d738, 0xfff15ca8, + 0x4eae9571, 0xff6b07e7, 0xf6cac3c7, 0x05da6dbe, 0xfeb2efcd, 0xff900089, 0x0057817b, 0xfff16a07, + 0x4e33e2ee, 0xfef898ae, 0xf6fae373, 0x05d6cb72, 0xfeaa34d0, 0xff957cf4, 0x005628ec, 0xfff17962, + 0x4db755f3, 0xfe87cc1b, 0xf72b425b, 0x05d2b05c, 0xfea1b5a9, 0xff9aede0, 0x0054cdc0, 0xfff18aab, + 0x4d38f520, 0xfe18a4bc, 0xf75bdbbd, 0x05ce1e2d, 0xfe997268, 0xffa052ec, 0x0053702d, 0xfff19dd1, + 0x4cb8c72e, 0xfdab2501, 0xf78caae0, 0x05c9169d, 0xfe916b15, 0xffa5abb8, 0x00521068, 0xfff1b2c5, + 0x4c36d2eb, 0xfd3f4f3d, 0xf7bdab16, 0x05c39b6a, 0xfe899fb2, 0xffaaf7e6, 0x0050aea5, 0xfff1c976, + 0x4bb31f3c, 0xfcd525a5, 0xf7eed7b4, 0x05bdae57, 0xfe82103f, 0xffb0371c, 0x004f4b17, 0xfff1e1d6, + 0x4b2db31a, 0xfc6caa53, 0xf8202c1c, 0x05b7512e, 0xfe7abcb1, 0xffb56902, 0x004de5f1, 0xfff1fbd5, + 0x4aa69594, 0xfc05df40, 0xf851a3b6, 0x05b085bc, 0xfe73a4fb, 0xffba8d44, 0x004c7f66, 0xfff21764, + 0x4a1dcdce, 0xfba0c64b, 0xf88339f5, 0x05a94dd5, 0xfe6cc909, 0xffbfa38d, 0x004b17a6, 0xfff23473, + 0x499362ff, 0xfb3d6133, 0xf8b4ea55, 0x05a1ab52, 0xfe6628c1, 0xffc4ab8f, 0x0049aee3, 0xfff252f3, + 0x49075c72, 0xfadbb19a, 0xf8e6b059, 0x0599a00e, 0xfe5fc405, 0xffc9a4fc, 0x0048454b, 0xfff272d6, + 0x4879c185, 0xfa7bb908, 0xf9188793, 0x05912dea, 0xfe599aaf, 0xffce8f8a, 0x0046db0f, 0xfff2940b, + 0x47ea99a9, 0xfa1d78e3, 0xf94a6b9b, 0x058856cd, 0xfe53ac97, 0xffd36af1, 0x0045705c, 0xfff2b686, + 0x4759ec60, 0xf9c0f276, 0xf97c5815, 0x057f1c9e, 0xfe4df98e, 0xffd836eb, 0x00440561, 0xfff2da36, + 0x46c7c140, 0xf96626f0, 0xf9ae48af, 0x0575814c, 0xfe48815e, 0xffdcf336, 0x00429a4a, 0xfff2ff0d, + 0x46341fed, 0xf90d1761, 0xf9e03924, 0x056b86c6, 0xfe4343d0, 0xffe19f91, 0x00412f43, 0xfff324fd, + 0x459f101d, 0xf8b5c4be, 0xfa122537, 0x05612f00, 0xfe3e40a6, 0xffe63bc0, 0x003fc478, 0xfff34bf9, + 0x45089996, 0xf8602fdc, 0xfa4408ba, 0x05567bf1, 0xfe39779a, 0xffeac787, 0x003e5a12, 0xfff373f0, + 0x4470c42d, 0xf80c5977, 0xfa75df87, 0x054b6f92, 0xfe34e867, 0xffef42af, 0x003cf03d, 0xfff39cd7, + 0x43d797c7, 0xf7ba422b, 0xfaa7a586, 0x05400be1, 0xfe3092bf, 0xfff3ad01, 0x003b871f, 0xfff3c69f, + 0x433d1c56, 0xf769ea78, 0xfad956ab, 0x053452dc, 0xfe2c7650, 0xfff8064b, 0x003a1ee3, 0xfff3f13a, + 0x42a159dc, 0xf71b52c4, 0xfb0aeef6, 0x05284685, 0xfe2892c5, 0xfffc4e5c, 0x0038b7ae, 0xfff41c9c, + 0x42045865, 0xf6ce7b57, 0xfb3c6a73, 0x051be8dd, 0xfe24e7c3, 0x00008507, 0x003751a7, 0xfff448b7, + 0x4166200e, 0xf683645a, 0xfb6dc53c, 0x050f3bec, 0xfe2174ec, 0x0004aa1f, 0x0035ecf4, 0xfff4757e, + 0x40c6b8fd, 0xf63a0ddf, 0xfb9efb77, 0x050241b6, 0xfe1e39da, 0x0008bd7c, 0x003489b9, 0xfff4a2e5, + 0x40262b65, 0xf5f277d9, 0xfbd00956, 0x04f4fc46, 0xfe1b3628, 0x000cbef7, 0x0033281a, 0xfff4d0de, + 0x3f847f83, 0xf5aca21f, 0xfc00eb1b, 0x04e76da3, 0xfe18696a, 0x0010ae6e, 0x0031c83a, 0xfff4ff5d, + 0x3ee1bda2, 0xf5688c6d, 0xfc319d13, 0x04d997d8, 0xfe15d32f, 0x00148bbd, 0x00306a3b, 0xfff52e57, + 0x3e3dee13, 0xf5263665, 0xfc621b9a, 0x04cb7cf2, 0xfe137304, 0x001856c7, 0x002f0e3f, 0xfff55dbf, + 0x3d991932, 0xf4e59f8a, 0xfc926319, 0x04bd1efb, 0xfe114872, 0x001c0f6e, 0x002db466, 0xfff58d89, + 0x3cf34766, 0xf4a6c748, 0xfcc27008, 0x04ae8000, 0xfe0f52fc, 0x001fb599, 0x002c5cd0, 0xfff5bdaa, + 0x3c4c811c, 0xf469aced, 0xfcf23eec, 0x049fa20f, 0xfe0d9224, 0x0023492f, 0x002b079a, 0xfff5ee17, + 0x3ba4cec9, 0xf42e4faf, 0xfd21cc59, 0x04908733, 0xfe0c0567, 0x0026ca1c, 0x0029b4e4, 0xfff61ec5, + 0x3afc38eb, 0xf3f4aea6, 0xfd5114f0, 0x0481317a, 0xfe0aac3f, 0x002a384c, 0x002864c9, 0xfff64fa8, + 0x3a52c805, 0xf3bcc8d3, 0xfd801564, 0x0471a2ef, 0xfe098622, 0x002d93ae, 0x00271766, 0xfff680b5, + 0x39a884a1, 0xf3869d1a, 0xfdaeca73, 0x0461dda0, 0xfe089283, 0x0030dc34, 0x0025ccd7, 0xfff6b1e4, + 0x38fd774e, 0xf3522a49, 0xfddd30eb, 0x0451e396, 0xfe07d0d3, 0x003411d2, 0x00248535, 0xfff6e329, + 0x3851a8a2, 0xf31f6f0f, 0xfe0b45aa, 0x0441b6dd, 0xfe07407d, 0x0037347d, 0x0023409a, 0xfff7147a, + 0x37a52135, 0xf2ee6a07, 0xfe39059b, 0x0431597d, 0xfe06e0eb, 0x003a442e, 0x0021ff1f, 0xfff745cd, + 0x36f7e9a4, 0xf2bf19ae, 0xfe666dbc, 0x0420cd80, 0xfe06b184, 0x003d40e0, 0x0020c0dc, 0xfff7771a, + 0x364a0a90, 0xf2917c6d, 0xfe937b15, 0x041014eb, 0xfe06b1ac, 0x00402a8e, 0x001f85e6, 0xfff7a857, + 0x359b8c9d, 0xf265908f, 0xfec02ac2, 0x03ff31c3, 0xfe06e0c4, 0x00430137, 0x001e4e56, 0xfff7d97a, + 0x34ec786f, 0xf23b544b, 0xfeec79ec, 0x03ee260d, 0xfe073e2a, 0x0045c4dd, 0x001d1a3f, 0xfff80a7c, + 0x343cd6af, 0xf212c5be, 0xff1865cd, 0x03dcf3ca, 0xfe07c93a, 0x00487582, 0x001be9b7, 0xfff83b52, + 0x338cb004, 0xf1ebe2ec, 0xff43ebac, 0x03cb9cf9, 0xfe08814e, 0x004b132b, 0x001abcd0, 0xfff86bf6, + 0x32dc0d17, 0xf1c6a9c3, 0xff6f08e4, 0x03ba2398, 0xfe0965bc, 0x004d9dde, 0x0019939d, 0xfff89c60, + 0x322af693, 0xf1a3181a, 0xff99badb, 0x03a889a1, 0xfe0a75da, 0x005015a5, 0x00186e31, 0xfff8cc86, + 0x3179751f, 0xf1812bb0, 0xffc3ff0c, 0x0396d10c, 0xfe0bb0f9, 0x00527a8a, 0x00174c9c, 0xfff8fc62, + 0x30c79163, 0xf160e22d, 0xffedd2fd, 0x0384fbd1, 0xfe0d166b, 0x0054cc9a, 0x00162eef, 0xfff92bec, + 0x30155404, 0xf1423924, 0x00173447, 0x03730be0, 0xfe0ea57e, 0x00570be4, 0x00151538, 0xfff95b1e, + 0x2f62c5a7, 0xf1252e0f, 0x00402092, 0x0361032a, 0xfe105d7e, 0x00593877, 0x0013ff88, 0xfff989ef, + 0x2eafeeed, 0xf109be56, 0x00689598, 0x034ee39b, 0xfe123db6, 0x005b5267, 0x0012edea, 0xfff9b85b, + 0x2dfcd873, 0xf0efe748, 0x0090911f, 0x033caf1d, 0xfe144570, 0x005d59c6, 0x0011e06d, 0xfff9e65a, + 0x2d498ad3, 0xf0d7a622, 0x00b81102, 0x032a6796, 0xfe1673f2, 0x005f4eac, 0x0010d71d, 0xfffa13e5, + 0x2c960ea3, 0xf0c0f808, 0x00df1328, 0x03180ee7, 0xfe18c884, 0x0061312e, 0x000fd205, 0xfffa40f8, + 0x2be26c73, 0xf0abda0e, 0x0105958c, 0x0305a6f0, 0xfe1b4268, 0x00630167, 0x000ed130, 0xfffa6d8d, + 0x2b2eaccf, 0xf0984931, 0x012b9635, 0x02f3318a, 0xfe1de0e2, 0x0064bf71, 0x000dd4a7, 0xfffa999d, + 0x2a7ad83c, 0xf086425a, 0x0151133e, 0x02e0b08d, 0xfe20a335, 0x00666b68, 0x000cdc74, 0xfffac525, + 0x29c6f738, 0xf075c260, 0x01760ad1, 0x02ce25ca, 0xfe2388a1, 0x0068056b, 0x000be89f, 0xfffaf01e, + 0x2913123c, 0xf066c606, 0x019a7b27, 0x02bb9310, 0xfe269065, 0x00698d98, 0x000af931, 0xfffb1a84, + 0x285f31b7, 0xf05949fb, 0x01be628c, 0x02a8fa2a, 0xfe29b9c1, 0x006b0411, 0x000a0e2f, 0xfffb4453, + 0x27ab5e12, 0xf04d4ade, 0x01e1bf58, 0x02965cdb, 0xfe2d03f2, 0x006c68f8, 0x000927a0, 0xfffb6d86, + 0x26f79fab, 0xf042c539, 0x02048ff8, 0x0283bce6, 0xfe306e35, 0x006dbc71, 0x00084589, 0xfffb961a, + 0x2643feda, 0xf039b587, 0x0226d2e6, 0x02711c05, 0xfe33f7c7, 0x006efea0, 0x000767f0, 0xfffbbe09, + 0x259083eb, 0xf032182f, 0x024886ad, 0x025e7bf0, 0xfe379fe3, 0x00702fae, 0x00068ed8, 0xfffbe552, + 0x24dd3721, 0xf02be98a, 0x0269a9e9, 0x024bde5a, 0xfe3b65c4, 0x00714fc0, 0x0005ba46, 0xfffc0bef, + 0x242a20b3, 0xf02725dc, 0x028a3b44, 0x023944ee, 0xfe3f48a5, 0x00725f02, 0x0004ea3a, 0xfffc31df, + 0x237748cf, 0xf023c95d, 0x02aa397b, 0x0226b156, 0xfe4347c0, 0x00735d9c, 0x00041eb9, 0xfffc571e, + 0x22c4b795, 0xf021d031, 0x02c9a359, 0x02142533, 0xfe476250, 0x00744bba, 0x000357c2, 0xfffc7ba9, + 0x2212751a, 0xf0213671, 0x02e877b9, 0x0201a223, 0xfe4b978e, 0x0075298a, 0x00029558, 0xfffc9f7e, + 0x21608968, 0xf021f823, 0x0306b586, 0x01ef29be, 0xfe4fe6b3, 0x0075f739, 0x0001d779, 0xfffcc29a, + 0x20aefc79, 0xf0241140, 0x03245bbc, 0x01dcbd96, 0xfe544efb, 0x0076b4f5, 0x00011e26, 0xfffce4fc, + 0x1ffdd63b, 0xf0277db1, 0x03416966, 0x01ca5f37, 0xfe58cf9d, 0x007762f0, 0x0000695e, 0xfffd06a1, + 0x1f4d1e8e, 0xf02c3953, 0x035ddd9e, 0x01b81028, 0xfe5d67d4, 0x0078015a, 0xffffb91f, 0xfffd2787, + 0x1e9cdd43, 0xf0323ff5, 0x0379b790, 0x01a5d1ea, 0xfe6216db, 0x00789065, 0xffff0d66, 0xfffd47ae, + 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631, 0xfffd6713, }; // we use 15 bits to interpolate between these samples @@ -96,12 +353,16 @@ void AudioResamplerSinc::init_routine() return; } - readResampleCoefficients = (readCoefficientsFn) dlsym(resampleCoeffLib, - "readResamplerCoefficients"); - readResampleFirNumCoeffFn readResampleFirNumCoeff = (readResampleFirNumCoeffFn) + readResampleFirNumCoeffFn readResampleFirNumCoeff; + readResampleFirLerpIntBitsFn readResampleFirLerpIntBits; + + readResampleCoefficients = (readCoefficientsFn) + dlsym(resampleCoeffLib, "readResamplerCoefficients"); + readResampleFirNumCoeff = (readResampleFirNumCoeffFn) dlsym(resampleCoeffLib, "readResampleFirNumCoeff"); - readResampleFirLerpIntBitsFn readResampleFirLerpIntBits = (readResampleFirLerpIntBitsFn) + readResampleFirLerpIntBits = (readResampleFirLerpIntBitsFn) dlsym(resampleCoeffLib, "readResampleFirLerpIntBits"); + if (!readResampleCoefficients || !readResampleFirNumCoeff || !readResampleFirLerpIntBits) { readResampleCoefficients = NULL; dlclose(resampleCoeffLib); @@ -111,15 +372,14 @@ void AudioResamplerSinc::init_routine() } c = &veryHighQualityConstants; - // we have 16 coefs samples per zero-crossing c->coefsBits = readResampleFirLerpIntBits(); - ALOGV("coefsBits = %d", c->coefsBits); c->cShift = kNumPhaseBits - c->coefsBits; c->cMask = ((1<<c->coefsBits)-1) << c->cShift; c->pShift = kNumPhaseBits - c->coefsBits - pLerpBits; c->pMask = ((1<<pLerpBits)-1) << c->pShift; // number of zero-crossing on each side c->halfNumCoefs = readResampleFirNumCoeff(); + ALOGV("coefsBits = %d", c->coefsBits); ALOGV("halfNumCoefs = %d", c->halfNumCoefs); // note that we "leak" resampleCoeffLib until the process exits } @@ -129,7 +389,7 @@ void AudioResamplerSinc::init_routine() static inline int32_t mulRL(int left, int32_t in, uint32_t vRL) { -#if defined(__arm__) && !defined(__thumb__) +#if USE_INLINE_ASSEMBLY int32_t out; if (left) { asm( "smultb %[out], %[in], %[vRL] \n" @@ -144,18 +404,15 @@ int32_t mulRL(int left, int32_t in, uint32_t vRL) } return out; #else - if (left) { - return int16_t(in>>16) * int16_t(vRL&0xFFFF); - } else { - return int16_t(in>>16) * int16_t(vRL>>16); - } + int16_t v = left ? int16_t(vRL) : int16_t(vRL>>16); + return int32_t((int64_t(in) * v) >> 16); #endif } static inline int32_t mulAdd(int16_t in, int32_t v, int32_t a) { -#if defined(__arm__) && !defined(__thumb__) +#if USE_INLINE_ASSEMBLY int32_t out; asm( "smlawb %[out], %[v], %[in], %[a] \n" : [out]"=r"(out) @@ -163,16 +420,14 @@ int32_t mulAdd(int16_t in, int32_t v, int32_t a) : ); return out; #else - return a + in * (v>>16); - // improved precision - // return a + in * (v>>16) + ((in * (v & 0xffff)) >> 16); + return a + int32_t((int64_t(v) * in) >> 16); #endif } static inline int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a) { -#if defined(__arm__) && !defined(__thumb__) +#if USE_INLINE_ASSEMBLY int32_t out; if (left) { asm( "smlawb %[out], %[v], %[inRL], %[a] \n" @@ -187,13 +442,8 @@ int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a) } return out; #else - if (left) { - return a + (int16_t(inRL&0xFFFF) * (v>>16)); - //improved precision - // return a + (int16_t(inRL&0xFFFF) * (v>>16)) + ((int16_t(inRL&0xFFFF) * (v & 0xffff)) >> 16); - } else { - return a + (int16_t(inRL>>16) * (v>>16)); - } + int16_t s = left ? int16_t(inRL) : int16_t(inRL>>16); + return a + int32_t((int64_t(v) * s) >> 16); #endif } @@ -202,7 +452,7 @@ int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a) AudioResamplerSinc::AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) : AudioResampler(bitDepth, inChannelCount, sampleRate, quality), - mState(0) + mState(0), mImpulse(0), mRingFull(0), mFirCoefs(0) { /* * Layout of the state buffer for 32 tap: @@ -220,44 +470,48 @@ AudioResamplerSinc::AudioResamplerSinc(int bitDepth, * */ + mVolumeSIMD[0] = 0; + mVolumeSIMD[1] = 0; + // Load the constants for coefficients int ok = pthread_once(&once_control, init_routine); if (ok != 0) { ALOGE("%s pthread_once failed: %d", __func__, ok); } - mConstants = (quality == VERY_HIGH_QUALITY) ? &veryHighQualityConstants : &highQualityConstants; + mConstants = (quality == VERY_HIGH_QUALITY) ? + &veryHighQualityConstants : &highQualityConstants; } -AudioResamplerSinc::~AudioResamplerSinc() -{ - delete[] mState; +AudioResamplerSinc::~AudioResamplerSinc() { + free(mState); } void AudioResamplerSinc::init() { - const Constants *c = mConstants; - - const size_t numCoefs = 2*c->halfNumCoefs; + const Constants& c(*mConstants); + const size_t numCoefs = 2 * c.halfNumCoefs; const size_t stateSize = numCoefs * mChannelCount * 2; - mState = new int16_t[stateSize]; + mState = (int16_t*)memalign(32, stateSize*sizeof(int16_t)); memset(mState, 0, sizeof(int16_t)*stateSize); - mImpulse = mState + (c->halfNumCoefs-1)*mChannelCount; + mImpulse = mState + (c.halfNumCoefs-1)*mChannelCount; mRingFull = mImpulse + (numCoefs+1)*mChannelCount; } +void AudioResamplerSinc::setVolume(int16_t left, int16_t right) { + AudioResampler::setVolume(left, right); + mVolumeSIMD[0] = int32_t(left)<<16; + mVolumeSIMD[1] = int32_t(right)<<16; +} + void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { - // FIXME store current state (up or down sample) and only load the coefs when the state // changes. Or load two pointers one for up and one for down in the init function. // Not critical now since the read functions are fast, but would be important if read was slow. if (mConstants == &veryHighQualityConstants && readResampleCoefficients) { - ALOGV("get coefficient from libmm-audio resampler library"); - mFirCoefs = (mInSampleRate <= mSampleRate) ? readResampleCoefficients(true) : - readResampleCoefficients(false); + mFirCoefs = readResampleCoefficients( mInSampleRate <= mSampleRate ); } else { - ALOGV("Use default coefficients"); mFirCoefs = (mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown; } @@ -270,7 +524,6 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, resample<2>(out, outFrameCount, provider); break; } - } @@ -278,7 +531,8 @@ template<int CHANNELS> void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { - const Constants *c = mConstants; + const Constants& c(*mConstants); + const size_t headOffset = c.halfNumCoefs*CHANNELS; int16_t* impulse = mImpulse; uint32_t vRL = mVolumeRL; size_t inputIndex = mInputIndex; @@ -313,43 +567,31 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, } } } - int16_t *in = mBuffer.i16; + int16_t const * const in = mBuffer.i16; const size_t frameCount = mBuffer.frameCount; // Always read-in the first samples from the input buffer - int16_t* head = impulse + c->halfNumCoefs*CHANNELS; - head[0] = in[inputIndex*CHANNELS + 0]; - if (CHANNELS == 2) - head[1] = in[inputIndex*CHANNELS + 1]; + int16_t* head = impulse + headOffset; + for (size_t i=0 ; i<CHANNELS ; i++) { + head[i] = in[inputIndex*CHANNELS + i]; + } // handle boundary case - int32_t l, r; - while (outputIndex < outputSampleCount) { - filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse); - out[outputIndex++] += 2 * mulRL(1, l, vRL); - out[outputIndex++] += 2 * mulRL(0, r, vRL); + while (CC_LIKELY(outputIndex < outputSampleCount)) { + filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL); + outputIndex += 2; phaseFraction += phaseIncrement; - const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; - if (phaseIndex == 1) { - inputIndex++; - if (inputIndex >= frameCount) - break; // need a new buffer - read<CHANNELS>(impulse, phaseFraction, in, inputIndex); - } else if (phaseIndex == 2) { // maximum value + const size_t phaseIndex = phaseFraction >> kNumPhaseBits; + for (size_t i=0 ; i<phaseIndex ; i++) { inputIndex++; - if (inputIndex >= frameCount) - break; // 0 frame available, 2 frames needed - // read first frame - read<CHANNELS>(impulse, phaseFraction, in, inputIndex); - inputIndex++; - if (inputIndex >= frameCount) - break; // 0 frame available, 1 frame needed - // read second frame + if (inputIndex >= frameCount) { + goto done; // need a new buffer + } read<CHANNELS>(impulse, phaseFraction, in, inputIndex); } } - +done: // if done with buffer, save samples if (inputIndex >= frameCount) { inputIndex -= frameCount; @@ -375,66 +617,215 @@ void AudioResamplerSinc::read( int16_t*& impulse, uint32_t& phaseFraction, const int16_t* in, size_t inputIndex) { - const Constants *c = mConstants; - const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; impulse += CHANNELS; phaseFraction -= 1LU<<kNumPhaseBits; - if (impulse >= mRingFull) { - const size_t stateSize = (c->halfNumCoefs*2)*CHANNELS; + + const Constants& c(*mConstants); + if (CC_UNLIKELY(impulse >= mRingFull)) { + const size_t stateSize = (c.halfNumCoefs*2)*CHANNELS; memcpy(mState, mState+stateSize, sizeof(int16_t)*stateSize); impulse -= stateSize; } - int16_t* head = impulse + c->halfNumCoefs*CHANNELS; - head[0] = in[inputIndex*CHANNELS + 0]; - if (CHANNELS == 2) - head[1] = in[inputIndex*CHANNELS + 1]; + + int16_t* head = impulse + c.halfNumCoefs*CHANNELS; + for (size_t i=0 ; i<CHANNELS ; i++) { + head[i] = in[inputIndex*CHANNELS + i]; + } } template<int CHANNELS> void AudioResamplerSinc::filterCoefficient( - int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples) + int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL) { - const Constants *c = mConstants; + // NOTE: be very careful when modifying the code here. register + // pressure is very high and a small change might cause the compiler + // to generate far less efficient code. + // Always sanity check the result with objdump or test-resample. // compute the index of the coefficient on the positive side and // negative side - uint32_t indexP = (phase & c->cMask) >> c->cShift; - uint16_t lerpP = (phase & c->pMask) >> c->pShift; - uint32_t indexN = (-phase & c->cMask) >> c->cShift; - uint16_t lerpN = (-phase & c->pMask) >> c->pShift; - if ((indexP == 0) && (lerpP == 0)) { - indexN = c->cMask >> c->cShift; - lerpN = c->pMask >> c->pShift; - } - - l = 0; - r = 0; - const int32_t* coefs = mFirCoefs; - const int16_t *sP = samples; - const int16_t *sN = samples+CHANNELS; - for (unsigned int i=0 ; i < c->halfNumCoefs/4 ; i++) { - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; + const Constants& c(*mConstants); + const int32_t ONE = c.cMask | c.pMask; + uint32_t indexP = ( phase & c.cMask) >> c.cShift; + uint32_t lerpP = ( phase & c.pMask) >> c.pShift; + uint32_t indexN = ((ONE-phase) & c.cMask) >> c.cShift; + uint32_t lerpN = ((ONE-phase) & c.pMask) >> c.pShift; + + const size_t offset = c.halfNumCoefs; + indexP *= offset; + indexN *= offset; + + int32_t const* coefsP = mFirCoefs + indexP; + int32_t const* coefsN = mFirCoefs + indexN; + int16_t const* sP = samples; + int16_t const* sN = samples + CHANNELS; + + size_t count = offset; + + if (!USE_NEON) { + int32_t l = 0; + int32_t r = 0; + for (size_t i=0 ; i<count ; i++) { + interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP); + sP -= CHANNELS; + interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN); + sN += CHANNELS; + } + out[0] += 2 * mulRL(1, l, vRL); + out[1] += 2 * mulRL(0, r, vRL); + } else if (CHANNELS == 1) { + int32_t const* coefsP1 = coefsP + offset; + int32_t const* coefsN1 = coefsN + offset; + sP -= CHANNELS*3; + asm ( + "vmov.32 d2[0], %[lerpP] \n" // load the positive phase + "vmov.32 d2[1], %[lerpN] \n" // load the negative phase + "veor q0, q0, q0 \n" // result, initialize to 0 + "vshl.s32 d2, d2, #16 \n" // convert to 32 bits + + "1: \n" + "vld1.16 { d4}, [%[sP]] \n" // load 4 16-bits stereo samples + "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs + "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation + "vld1.16 { d6}, [%[sN]]! \n" // load 4 16-bits stereo samples + "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs + "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation + + "vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side + + "vsub.s32 q9, q9, q8 \n" // interpolate (step1) 1st set of coefs + "vsub.s32 q11, q11, q10 \n" // interpolate (step1) 2nd set of coets + "vshll.s16 q12, d4, #15 \n" // extend samples to 31 bits + + "vqrdmulh.s32 q9, q9, d2[0] \n" // interpolate (step2) 1st set of coefs + "vqrdmulh.s32 q11, q11, d2[1] \n" // interpolate (step3) 2nd set of coefs + "vshll.s16 q14, d6, #15 \n" // extend samples to 31 bits + + "vadd.s32 q8, q8, q9 \n" // interpolate (step3) 1st set + "vadd.s32 q10, q10, q11 \n" // interpolate (step4) 2nd set + "subs %[count], %[count], #4 \n" // update loop counter + + "vqrdmulh.s32 q12, q12, q8 \n" // multiply samples by interpolated coef + "vqrdmulh.s32 q14, q14, q10 \n" // multiply samples by interpolated coef + "sub %[sP], %[sP], #8 \n" // move pointer to next set of samples + + "vadd.s32 q0, q0, q12 \n" // accumulate result + "vadd.s32 q0, q0, q14 \n" // accumulate result + + "bne 1b \n" // loop + + "vld1.s32 {d2}, [%[vLR]] \n" // load volumes + "vld1.s32 {d3}, %[out] \n" // load the output + "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums + "vpadd.s32 d0, d0, d0 \n" // together + "vdup.i32 d0, d0[0] \n" // interleave L,R channels + "vqrdmulh.s32 d0, d0, d2 \n" // apply volume + "vadd.s32 d3, d3, d0 \n" // accumulate result + "vst1.s32 {d0}, %[out] \n" // store result + + : [out] "=Uv" (out[0]), + [count] "+r" (count), + [coefsP0] "+r" (coefsP), + [coefsP1] "+r" (coefsP1), + [coefsN0] "+r" (coefsN), + [coefsN1] "+r" (coefsN1), + [sP] "+r" (sP), + [sN] "+r" (sN) + : [lerpP] "r" (lerpP), + [lerpN] "r" (lerpN), + [vLR] "r" (mVolumeSIMD) + : "cc", "memory", + "q0", "q1", "q2", "q3", + "q8", "q9", "q10", "q11", + "q12", "q14" + ); + } else if (CHANNELS == 2) { + int32_t const* coefsP1 = coefsP + offset; + int32_t const* coefsN1 = coefsN + offset; + sP -= CHANNELS*3; + asm ( + "vmov.32 d2[0], %[lerpP] \n" // load the positive phase + "vmov.32 d2[1], %[lerpN] \n" // load the negative phase + "veor q0, q0, q0 \n" // result, initialize to 0 + "veor q4, q4, q4 \n" // result, initialize to 0 + "vshl.s32 d2, d2, #16 \n" // convert to 32 bits + + "1: \n" + "vld2.16 {d4,d5}, [%[sP]] \n" // load 4 16-bits stereo samples + "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs + "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation + "vld2.16 {d6,d7}, [%[sN]]! \n" // load 4 16-bits stereo samples + "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs + "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation + + "vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side + "vrev64.16 d5, d5 \n" // reverse 2 frames of the positive side + + "vsub.s32 q9, q9, q8 \n" // interpolate (step1) 1st set of coefs + "vsub.s32 q11, q11, q10 \n" // interpolate (step1) 2nd set of coets + "vshll.s16 q12, d4, #15 \n" // extend samples to 31 bits + "vshll.s16 q13, d5, #15 \n" // extend samples to 31 bits + + "vqrdmulh.s32 q9, q9, d2[0] \n" // interpolate (step2) 1st set of coefs + "vqrdmulh.s32 q11, q11, d2[1] \n" // interpolate (step3) 2nd set of coefs + "vshll.s16 q14, d6, #15 \n" // extend samples to 31 bits + "vshll.s16 q15, d7, #15 \n" // extend samples to 31 bits + + "vadd.s32 q8, q8, q9 \n" // interpolate (step3) 1st set + "vadd.s32 q10, q10, q11 \n" // interpolate (step4) 2nd set + "subs %[count], %[count], #4 \n" // update loop counter + + "vqrdmulh.s32 q12, q12, q8 \n" // multiply samples by interpolated coef + "vqrdmulh.s32 q13, q13, q8 \n" // multiply samples by interpolated coef + "vqrdmulh.s32 q14, q14, q10 \n" // multiply samples by interpolated coef + "vqrdmulh.s32 q15, q15, q10 \n" // multiply samples by interpolated coef + "sub %[sP], %[sP], #16 \n" // move pointer to next set of samples + + "vadd.s32 q0, q0, q12 \n" // accumulate result + "vadd.s32 q4, q4, q13 \n" // accumulate result + "vadd.s32 q0, q0, q14 \n" // accumulate result + "vadd.s32 q4, q4, q15 \n" // accumulate result + + "bne 1b \n" // loop + + "vld1.s32 {d2}, [%[vLR]] \n" // load volumes + "vld1.s32 {d3}, %[out] \n" // load the output + "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums from q0 + "vpadd.s32 d8, d8, d9 \n" // add all 4 partial sums from q4 + "vpadd.s32 d0, d0, d0 \n" // together + "vpadd.s32 d8, d8, d8 \n" // together + "vtrn.s32 d0, d8 \n" // interlace L,R channels + "vqrdmulh.s32 d0, d0, d2 \n" // apply volume + "vadd.s32 d3, d3, d0 \n" // accumulate result + "vst1.s32 {d0}, %[out] \n" // store result + + : [out] "=Uv" (out[0]), + [count] "+r" (count), + [coefsP0] "+r" (coefsP), + [coefsP1] "+r" (coefsP1), + [coefsN0] "+r" (coefsN), + [coefsN1] "+r" (coefsN1), + [sP] "+r" (sP), + [sN] "+r" (sN) + : [lerpP] "r" (lerpP), + [lerpN] "r" (lerpN), + [vLR] "r" (mVolumeSIMD) + : "cc", "memory", + "q0", "q1", "q2", "q3", "q4", + "q8", "q9", "q10", "q11", + "q12", "q13", "q14", "q15" + ); } } template<int CHANNELS> void AudioResamplerSinc::interpolate( int32_t& l, int32_t& r, - const int32_t* coefs, int16_t lerp, const int16_t* samples) + const int32_t* coefs, size_t offset, + int32_t lerp, const int16_t* samples) { int32_t c0 = coefs[0]; - int32_t c1 = coefs[1]; + int32_t c1 = coefs[offset]; int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0); if (CHANNELS == 2) { uint32_t rl = *reinterpret_cast<const uint32_t*>(samples); diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h index 25fc025..09c6866 100644 --- a/services/audioflinger/AudioResamplerSinc.h +++ b/services/audioflinger/AudioResamplerSinc.h @@ -44,18 +44,21 @@ public: private: void init(); + virtual void setVolume(int16_t left, int16_t right); + template<int CHANNELS> void resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); template<int CHANNELS> inline void filterCoefficient( - int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples); + int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL); template<int CHANNELS> inline void interpolate( int32_t& l, int32_t& r, - const int32_t* coefs, int16_t lerp, const int16_t* samples); + const int32_t* coefs, size_t offset, + int32_t lerp, const int16_t* samples); template<int CHANNELS> inline void read(int16_t*& impulse, uint32_t& phaseFraction, @@ -64,6 +67,7 @@ private: int16_t *mState; int16_t *mImpulse; int16_t *mRingFull; + int32_t mVolumeSIMD[2]; const int32_t * mFirCoefs; static const int32_t mFirCoefsDown[]; @@ -71,17 +75,14 @@ private: // ---------------------------------------------------------------------------- static const int32_t RESAMPLE_FIR_NUM_COEF = 8; - static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4; + static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 7; struct Constants { - // we have 16 coefs samples per zero-crossing int coefsBits; int cShift; uint32_t cMask; - int pShift; uint32_t pMask; - // number of zero-crossing on each side unsigned int halfNumCoefs; }; diff --git a/services/audioflinger/StateQueue.h b/services/audioflinger/StateQueue.h index eba190c..e33b3c6 100644 --- a/services/audioflinger/StateQueue.h +++ b/services/audioflinger/StateQueue.h @@ -17,6 +17,72 @@ #ifndef ANDROID_AUDIO_STATE_QUEUE_H #define ANDROID_AUDIO_STATE_QUEUE_H +// The state queue template class was originally driven by this use case / requirements: +// There are two threads: a fast mixer, and a normal mixer, and they share state. +// The interesting part of the shared state is a set of active fast tracks, +// and the output HAL configuration (buffer size in frames, sample rate, etc.). +// Fast mixer thread: +// periodic with typical period < 10 ms +// FIFO/RR scheduling policy and a low fixed priority +// ok to block for bounded time using nanosleep() to achieve desired period +// must not block on condition wait, mutex lock, atomic operation spin, I/O, etc. +// under typical operations of mixing, writing, or adding/removing tracks +// ok to block for unbounded time when the output HAL configuration changes, +// and this may result in an audible artifact +// needs read-only access to a recent stable state, +// but not necessarily the most current one +// Normal mixer thread: +// periodic with typical period ~40 ms +// SCHED_OTHER scheduling policy and nice priority == urgent audio +// ok to block, but prefer to avoid as much as possible +// needs read/write access to state +// The normal mixer may need to temporarily suspend the fast mixer thread during mode changes. +// It will do this using the state -- one of the fields tells the fast mixer to idle. + +// Additional requirements: +// - observer must always be able to poll for and view the latest pushed state; it must never be +// blocked from seeing that state +// - observer does not need to see every state in sequence; it is OK for it to skip states +// [see below for more on this] +// - mutator must always be able to read/modify a state, it must never be blocked from reading or +// modifying state +// - reduce memcpy where possible +// - work well if the observer runs more frequently than the mutator, +// as is the case with fast mixer/normal mixer. +// It is not a requirement to work well if the roles were reversed, +// and the mutator were to run more frequently than the observer. +// In this case, the mutator could get blocked waiting for a slot to fill up for +// it to work with. This could be solved somewhat by increasing the depth of the queue, but it would +// still limit the mutator to a finite number of changes before it would block. A future +// possibility, not implemented here, would be to allow the mutator to safely overwrite an already +// pushed state. This could be done by the mutator overwriting mNext, but then being prepared to +// read an mAck which is actually for the earlier mNext (since there is a race). + +// Solution: +// Let's call the fast mixer thread the "observer" and normal mixer thread the "mutator". +// We assume there is only a single observer and a single mutator; this is critical. +// Each state is of type <T>, and should contain only POD (Plain Old Data) and raw pointers, as +// memcpy() may be used to copy state, and the destructors are run in unpredictable order. +// The states in chronological order are: previous, current, next, and mutating: +// previous read-only, observer can compare vs. current to see the subset that changed +// current read-only, this is the primary state for observer +// next read-only, when observer is ready to accept a new state it will shift it in: +// previous = current +// current = next +// and the slot formerly used by previous is now available to the mutator. +// mutating invisible to observer, read/write to mutator +// Initialization is tricky, especially for the observer. If the observer starts execution +// before the mutator, there are no previous, current, or next states. And even if the observer +// starts execution after the mutator, there is a next state but no previous or current states. +// To solve this, we'll have the observer idle until there is a next state, +// and it will have to deal with the case where there is no previous state. +// The states are stored in a shared FIFO queue represented using a circular array. +// The observer polls for mutations, and receives a new state pointer after a +// a mutation is pushed onto the queue. To the observer, the state pointers are +// effectively in random order, that is the observer should not do address +// arithmetic on the state pointers. However to the mutator, the state pointers +// are in a definite circular order. + namespace android { #ifdef STATE_QUEUE_DUMP @@ -108,7 +174,7 @@ public: #endif private: - static const unsigned kN = 4; // values != 4 are not supported by this code + static const unsigned kN = 4; // values < 4 are not supported by this code T mStates[kN]; // written by mutator, read by observer // "volatile" is meaningless with SMP, but here it indicates that we're using atomic ops diff --git a/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp b/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp index ade58a7..af3e40d 100644 --- a/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp +++ b/services/audioflinger/audio-resampler/AudioResamplerCoefficients.cpp @@ -14,42 +14,41 @@ * limitations under the License. */ -#include <dnsampler_filter_coefficients_x128_10112011.h> -#include <resampler_filter_coefficients_10042011.h> -#undef LOG_TAG -#include <utils/Log.h> -//#include "common_log.h" #define LOG_TAG "ResamplerCoefficients" #define LOG_NDEBUG 0 -const int32_t RESAMPLE_FIR_NUM_COEF = 16; -const int32_t RESAMPLE_FIR_LERP_INT_BITS = 7; +#include <utils/Log.h> + +#include "filter_coefficients.h" + +const int32_t RESAMPLE_FIR_NUM_COEF = 16; +const int32_t RESAMPLE_FIR_LERP_INT_BITS = 7; using namespace android; + #ifdef __cplusplus extern "C" { #endif + const int32_t* readResamplerCoefficients(bool upSample) { ALOGV("readResamplerCoefficients"); - if(upSample) { - return resampler_filter_coefficients_10042011; + if (upSample) { + return up_sampler_filter_coefficients; + } else { + return dn_sampler_filter_coefficients; } - else { - return dnsampler_filter_coefficients_x128_10112011; - } } int32_t readResampleFirNumCoeff() { - return RESAMPLE_FIR_NUM_COEF; } int32_t readResampleFirLerpIntBits() { - - return RESAMPLE_FIR_LERP_INT_BITS; + return RESAMPLE_FIR_LERP_INT_BITS; } + #ifdef __cplusplus } #endif diff --git a/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h b/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h deleted file mode 100644 index eb2944c..0000000 --- a/services/audioflinger/audio-resampler/dnsampler_filter_coefficients_x128_10112011.h +++ /dev/null @@ -1,2585 +0,0 @@ - -/* - * Copyright (C) 2012 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#include <stdlib.h> - -namespace android { - -const int32_t dnsampler_filter_coefficients_x128_10112011[] = { -1849391518, -1849249650, -1848824221, -1848115177, -1847122891, -1845847499, -1844289491, -1842449129, -1840327103, -1837923861, -1835240203, -1832276710, -1829034388, -1825513999, -1821716652, -1817643240, -1813295074, -1808673214, -1803779065, -1798613825, -1793179109, -1787476282, -1781507048, -1775272909, -1768775774, -1762017295, -1754999460, -1747724062, -1740193297, -1732409109, -1724373764, -1716089338, -1707558293, -1698782831, -1689765473, -1680508558, -1671014814, -1661286715, -1651327042, -1641138399, -1630723762, -1620085842, -1609227651, -1598152036, -1586862206, -1575361116, -1563652006, -1551737947, -1539622358, -1527308388, -1514799466, -1502098869, -1489210218, -1476136877, -1462882477, -1449450498, -1435844739, -1422068735, -1408126274, -1394021008, -1379756900, -1365337657, -1350767225, -1336049408, -1321188297, -1306187721, -1291051731, -1275784259, -1260389523, -1244871491, -1229234343, -1213482137, -1197619187, -1181649550, -1165577481, -1149407125, -1133142876, -1116788884, -1100349476, -1083828868, -1067231493, -1050561533, -1033823337, -1017021160, -1000159475, -983242522, -966274693, -949260282, -932203771, -915109401, -897981548, -880824504, -863642743, -846440509, -829222164, -811991981, -794754382, -777513558, -760273800, -743039333, -725814532, -708603558, -691410662, -674240023, -657095935, -639982476, -622903794, -605863979, -588867231, -571917545, -555018972, -538175499, -521391190, -504669905, -488015541, -471431958, -454923090, -438492688, -422144531, -405882352, -389709927, -373630848, -357648715, -341767096, -325989596, -310319649, -294760687, -279316098, -263989284, -248783473, -233701878, -218747691, -203924116, -189234208, -174680993, -160267473, -145996632, -131871305, -117894282, -104068338, -90396230, -76880575, -63523937, -50328860, -37297842, -24433248, -11737378, --787473, --13139049, --25315210, --37313887, --49133021, 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-3961262, -3709774, -3458055, -3206216, -2954367, -2702617, -2451073, -2199845, -1949039, -1698760, -1449115, -1200206, -952139, -705015, -458936, -214002, --29687, --272033, --512941, --752313, --990055, --1226074, --1460278, --1692576, --1922878, --2151097, --2377147, --2600942, --2822399, --3041435, --3257970, --3471925, --3683223, --3891787, --4097545, --4300424, --4500352, --4697261, --4891083, --5081754, --5269208, --5453384, --5634222, --5811664, --5985653, --6156134, --6323055, --6486364, --6646013, --6801954, --6954144, --7102537, --7247093, --7387773, --7524538, --7657353, --7786184, --7910999, --8031770, --8148467, --8261064, --8369539, --8473868, --8574031, --8670011, --8761790, --8849356, --8932694, --9011796, --9086651, --9157254, --9223598, --9285682, --9343504, --9397065, --9446367, --9491415, --9532214, --9568774, --9601103, --9629214, --9653119, --9672835, --9688376, --9699763, --9707015, --9710154, --9709202, --9704187, --9695132, --9682068, --9665024, --9644031, --9619121, --9590330, --9557692, --9521245, --9481028, --9437080, --9389443, --9338159, --9283272, --9224828, --9162872, --9097452, --9028617, --8956417, --8880903, --8802127, --8720142, --8635003, --8546764, --8455484, --8361217, --8264022, --8163959, --8061088, --7955468, --7847161, --7736229, --7622737, --7506745, --7388320, --7267525, --7144428, --7019092, --6891585, --6761974, --6630327, --6496711, --6361194, --6223845, --6084734, --5943929, --5801500, --5657517, --5512050, --5365168, --5216942, --5067443, --4916741, --4764906, --4612009, --4458120, --4303311, --4147650, --3991209, --3834057, --3676265, --3517901, --3359037, --3199740, --3040080, --2880126, --2719945, --2559606, --2399177, --2238724, --2078315, --1918015, --1757891, --1598007, --1438428, --1279218, --1120441, --962159, --804435, --647330, --490906, --335223, --180338, --26312, -126797, -278933, -430040, -580062, -728943, -876632, -1023074, -1168219, -1312015, -1454411, -1595360, -1734812, -1872720, -2009038, -2143721, -2276725, -2408006, -2537522, -2665233, -2791097, -2915076, -3037131, -3157228, -3275328, -3391399, -3505405, -3617315, -3727097, -3834720, -3940156, -4043377, -4144356, -4243067, -4339485, -4433587, -4525351, -4614755, -4701780, -4786406, -4868616, -4948393, -5025721, -5100587, -5172977, -5242878, -5310279, -5375172, -5437547, -5497396, -5554712, -5609491, -5661728, -5711419, -5758562, -5803157, -5845203, -5884701, -5921652, -5956061, -5987931, -6017267, -6044075, -6068363, -6090138, -6109409, -6126186, -6140482, -6152306, -6161673, -6168595, -6173088, -6175167, -6174849, -6172151, -6167091, -6159687, -6149960, -6137930, -6123618, -6107046, -6088238, -6067217, -6044007, -6018632, -5991120, -5961496, -5929787, -5896020, -5860225, -5822429, -5782663, -5740956, -5697338, -5651841, -5604497, -5555336, -5504393, -5451699, -5397289, -5341196, -5283454, -5224099, -5163164, -5100686, -5036700, -4971242, -4904348, -4836055, -4766400, -4695420, -4623152, -4549633, -4474901, -4398995, -4321952, -4243810, -4164608, -4084383, -4003175, -3921021, -3837960, -3754032, -3669273, -3583724, -3497422, -3410406, -3322714, -3234384, -3145456, -3055967, -2965954, -2875457, -2784513, -2693159, -2601433, -2509371, -2417012, -2324391, -2231545, -2138511, -2045325, -1952022, -1858638, -1765208, -1671768, -1578350, -1484991, -1391723, -1298581, -1205597, -1112804, -1020235, -927921, -835894, -744186, -652826, -561846, -471276, -381143, -291479, -202310, -113666, -25572, --61942, --148851, --235129, --320751, --405691, --489926, --573430, --656181, --738156, --819332, --899688, --979202, --1057854, --1135623, --1212489, --1288435, --1363441, --1437489, --1510563, --1582644, --1653718, --1723769, --1792781, --1860740, --1927631, --1993443, --2058161, --2121774, --2184270, --2245638, --2305868, --2364949, --2422873, --2479630, --2535212, --2589613, --2642824, --2694840, --2745654, --2795261, --2843656, --2890834, --2936793, --2981528, --3025036, --3067316, --3108366, --3148185, --3186772, --3224126, --3260249, --3295140, --3328802, --3361236, --3392444, --3422430, --3451195, --3478744, --3505081, --3530210, --3554136, --3576865, --3598402, --3618754, --3637927, --3655929, --3672765, --3688446, --3702978, --3716370, --3728631, --3739770, --3749797, --3758721, --3766553, --3773304, --3778983, --3783603, --3787174, --3789709, --3791219, --3791717, --3791215, --3789725, --3787262, --3783838, --3779466, --3774160, --3767934, --3760802, --3752778, --3743877, --3734113, --3723501, --3712056, --3699792, --3686725, --3672871, --3658244, --3642860, --3626735, --3609884, --3592324, --3574069, --3555137, --3535542, --3515302, --3494432, --3472947, --3450866, --3428203, --3404974, --3381197, --3356887, --3332060, --3306732, --3280921, --3254641, --3227909, --3200741, --3173153, --3145162, --3116781, --3088029, --3058920, --3029469, --2999694, --2969608, --2939228, --2908568, --2877644, --2846472, --2815064, --2783438, --2751607, --2719586, --2687389, --2655030, --2622524, --2589885, --2557126, --2524261, --2491304, --2458268, --2425166, --2392011, --2358815, --2325592, --2292354, --2259114, --2225882, --2192671, --2159493, --2126359, --2093281, --2060269, --2027334, --1994487, --1961738, --1929098, --1896576, --1864183, --1831927, --1799818, --1767865, --1736078, --1704464, --1673032, --1641791, --1610748, --1579911, --1549288, --1518886, --1488713, --1458774, --1429078, --1399630, --1370436, --1341503, --1312836, --1284441, --1256324, --1228489, --1200941, --1173686, --1146727, --1120069, --1093717, --1067673, --1041942, --1016527, --991432, --966659, --942211, --918092, --894303, --870846, --847724, --824940, --802493, --780386, --758621, --737199, --716120, --695385, --674996, --654951, --635253, --615901, --596895, --578234, --559919, --541949, --524324, --507042, --490103, --473505, --457248, --441330, --425750, --410506, --395596, --381019, --366773, --352855, --339263, --325995, --313049, --300421, --288110, --276112, --264425, --253046, --241972, --231199, --220725, --210546, --200659, --191060, --181746, --172713, --163958, --155477, --147266, --139321, --131639, --124215, --117045, --110126, --103453, --97022, --90829, --84870, --79140, --73636, --68352, --63285, --58431, --53784, --49341, --45097, --41048, --37188, --33515, --30023, --26708, --23566, --20592, --17782, --15130, --12634, --10289, --8089, --6031, --4111, --2324, --666, -868, -2281, -3578, -4763, -5839, -6812, -7685, -8462, -9146, -9743, -10255, -10686, -11041, -11322, -11533, -11679, -11762, -11785, -11753, -11668, -11533, -11352, -11128, -10864, -10563, -10227, -9860, -9464, -9042, -8596, -8129, -7644, -7142, -6626, -6099, -5562, -5017, -4467, -3913, -3357, -2800, -2245, -1694, -1146, -605, -71, --454, --970, --1474, --1966, --2446, --2911, --3362, --3797, --4215, --4617, --5001, --5367, --5714, --6043, --6352, --6641, --6910, --7160, --7389, --7598, --7786, --7954, --8102, --8230, --8338, --8426, --8495, --8545, --8575, --8587, --8582, --8558, --8517, --8460, --8386, --8297, --8192, --8073, --7940, --7794, --7635, --7464, --7281, --7088, --6885, --6672, --6450, --6221, --5984, --5740, --5490, --5235, --4975, --4711, --4443, --4173, --3901, --3627, --3352, --3077, --2803, --2529, --2257, --1986, --1719, --1454, --1193, --935, --683, --435, --192, -45, -276, -501, -719, -930, -1134, -1331, -1519, -1700, -1873, -2038, -2194, -2342, -2481, -2611, -2733, -2846, -2950, -3046, -3133, -3211, -3281, -3343, -3396, -3441, -3477, -3506, -3527, -3541, -3547, -3546, -3538, -3523, -3502, -3474, -3441, -3401, -3357, -3307, -3252, -3192, -3128, -3060, -2989, -2913, -2835, -2753, -2669, -2583, -2494, -2403, -2311, -2218, -2124, -2028, -1933, -1837, -1741, -1645, -1550, -1455, -1361, -1268, -1176, -1086, -997, -910, -825, -741, -660, -581, -504, -429, -357, -287, -220, -156, -94, -35, --22, --75, --126, --175, --220, --263, --303, --341, --375, --408, --437, --464, --489, --511, --531, --548, --564, --577, --588, --597, --604, --610, --613, --615, --616, --614, --612, --608, --603, --597, --590, --582, --573, --563, --553, --542, --530, --518, --506, --493, --480, --466, --453, --439, --425, --411, --398, --384, --370, --357, --344, --331, --318, --305, --293, --281, --269, --258, --247, --237, --227, --217, --208, --199, --190, --182, --174, --167, --160, --154, --147, --142, --136, --131, --126, --121, --117, --113, --109, --106, --102, --99, --96, --93, --90, --87, --85, --82, --80, --78, --76, --74, --72, --70, --68, --66, --64, --62, --60, --58, --57, --55, --53, --51, --50, --48, --46, --45, --43, --41, --40, --38, --36, --35, --33, --31, --30, --28, --27, --25, --24, --22, --21, --20, --18, --17, --16, --15, --13, --12, --11, --10, --9, --9, --8, --7, --6, -}; -} diff --git a/services/audioflinger/audio-resampler/filter_coefficients.h b/services/audioflinger/audio-resampler/filter_coefficients.h new file mode 100644 index 0000000..bf70c63 --- /dev/null +++ b/services/audioflinger/audio-resampler/filter_coefficients.h @@ -0,0 +1,285 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ +#include <stdlib.h> + +namespace android { + +// cmd-line: fir -l 7 -s 48000 -c 23400 -n 16 -b 9.62 +const int32_t up_sampler_filter_coefficients[] __attribute__ ((aligned (32))) = { + 0x7ccccccd, 0x0323eb7f, 0xfd086246, 0x02b2aa5c, 0xfda45e2c, 0x01fa5183, 0xfe694e12, 0x0137e672, 0xff1c87d3, 0x009ce6d8, 0xff9a68b0, 0x003d150d, 0xffde727a, 0x00106595, 0xfff93679, 0x00021fc5, + 0x7cc9b757, 0x022ac835, 0xfd7e3a71, 0x026b7da1, 0xfdd2b905, 0x01db7c90, 0xfe7db77c, 0x012aa7bf, 0xff24dc32, 0x0097dfc9, 0xff9d4ae9, 0x003b8742, 0xffdf38e5, 0x00100be5, 0xfff959f5, 0x0002144b, + 0x7cc0773c, 0x01354bc1, 0xfdf365e8, 0x0224726d, 0xfe011d2e, 0x01bc908b, 0xfe923a2b, 0x011d528d, 0xff2d426f, 0x0092cbc0, 0xffa035cc, 0x0039f42e, 0xffe00236, 0x000fb0d2, 0xfff97dfa, 0x000208b0, + 0x7cb10d52, 0x0043843f, 0xfe67d5a8, 0x01dd92df, 0xfe2f83c1, 0x019d9230, 0xfea6d2e5, 0x010fe901, 0xff35b924, 0x008dab9d, 0xffa328d4, 0x00385c1d, 0xffe0ce46, 0x000f5471, 0xfff9a27f, 0x0001fcf5, + 0x7c9b7afd, 0xff557f58, 0xfedb7ae9, 0x0196e8fe, 0xfe5de5e3, 0x017e8635, 0xfebb7e75, 0x01026d40, 0xff3e3eed, 0x0088803e, 0xffa6237a, 0x0036bf58, 0xffe19cec, 0x000ef6d4, 0xfff9c77d, 0x0001f11e, + 0x7c7fc22f, 0xfe6b4a44, 0xff4e471d, 0x01507eb8, 0xfe8c3cc3, 0x015f714d, 0xfed039a8, 0x00f4e16f, 0xff46d266, 0x00834a83, 0xffa9253b, 0x00351e2d, 0xffe26e01, 0x000e980f, 0xfff9eceb, 0x0001e52e, + 0x7c5de56a, 0xfd84f1c8, 0xffc02bf2, 0x010a5de2, 0xfeba819d, 0x01405821, 0xfee5014c, 0x00e747b0, 0xff4f722b, 0x007e0b4b, 0xffac2d8f, 0x003378e7, 0xffe3415d, 0x000e3834, 0xfffa12c0, 0x0001d927, + 0x7c35e7bb, 0xfca28234, 0x00311b54, 0x00c49034, 0xfee8adba, 0x01213f58, 0xfef9d232, 0x00d9a226, 0xff581cd8, 0x0078c375, 0xffaf3bf2, 0x0031cfd1, 0xffe416d8, 0x000dd758, 0xfffa38f5, 0x0001cd0d, + 0x7c07ccbe, 0xfbc40766, 0x00a1076e, 0x007f1f4b, 0xff16ba71, 0x01022b90, 0xff0ea931, 0x00cbf2f0, 0xff60d10b, 0x007373de, 0xffb24fde, 0x00302337, 0xffe4ee4b, 0x000d758d, 0xfffa5f81, 0x0001c0e1, + 0x7bd3989d, 0xfae98cc5, 0x010fe2ab, 0x003a14a6, 0xff44a128, 0x00e3215e, 0xff238322, 0x00be3c2d, 0xff698d62, 0x006e1d66, 0xffb568ce, 0x002e7363, 0xffe5c78d, 0x000d12e6, 0xfffa865d, 0x0001b4a8, + 0x7b99500c, 0xfa131d41, 0x017d9fb8, 0xfff579a3, 0xff725b54, 0x00c42551, 0xff385ce3, 0x00b07ff8, 0xff72507e, 0x0068c0e9, 0xffb8863e, 0x002cc0a2, 0xffe6a277, 0x000caf76, 0xfffaad81, 0x0001a863, + 0x7b58f84d, 0xf940c355, 0x01ea3184, 0xffb15783, 0xff9fe27d, 0x00a53bed, 0xff4d3358, 0x00a2c06b, 0xff7b18fe, 0x00635f45, 0xffbba7aa, 0x002b0b3d, 0xffe77ee2, 0x000c4b50, 0xfffad4e4, 0x00019c15, + 0x7b12972d, 0xf8728902, 0x02558b43, 0xff6db764, 0xffcd303b, 0x008669ae, 0xff620368, 0x0094ff9b, 0xff83e586, 0x005df954, 0xffbecc8d, 0x00295380, 0xffe85ca7, 0x000be687, 0xfffafc7f, 0x00018fc1, + 0x7ac63304, 0xf7a877d4, 0x02bfa06d, 0xff2aa243, 0xfffa3e37, 0x0067b303, 0xff76ca02, 0x00873f9b, 0xff8cb4bb, 0x00588ff1, 0xffc1f465, 0x002799b3, 0xffe93b9e, 0x000b812d, 0xfffb244a, 0x0001836a, + 0x7a73d2b5, 0xf6e298db, 0x032864c1, 0xfee820f8, 0x00270631, 0x00491c54, 0xff8b841a, 0x0079827a, 0xff958542, 0x005323f7, 0xffc51eaf, 0x0025de22, 0xffea1ba2, 0x000b1b55, 0xfffb4c3e, 0x00017712, + 0x7a1b7daa, 0xf620f4b2, 0x038fcc44, 0xfea63c38, 0x005381fa, 0x002aa9fa, 0xffa02eac, 0x006bca44, 0xff9e55c6, 0x004db63c, 0xffc84ae9, 0x00242115, 0xffeafc8b, 0x000ab510, 0xfffb7452, 0x00016abb, + 0x79bd3bd8, 0xf5639376, 0x03f5cb46, 0xfe64fc93, 0x007fab77, 0x000c6043, 0xffb4c6b9, 0x005e1900, 0xffa724f0, 0x00484799, 0xffcb7893, 0x002262d6, 0xffebde33, 0x000a4e72, 0xfffb9c80, 0x00015e68, + 0x795915bc, 0xf4aa7cce, 0x045a565c, 0xfe246a72, 0x00ab7ca6, 0xffee4372, 0xffc9494b, 0x005070b0, 0xffaff16f, 0x0042d8e1, 0xffcea72c, 0x0020a3ad, 0xffecc075, 0x0009e78c, 0xfffbc4bf, 0x0001521b, + 0x78ef1457, 0xf3f5b7e4, 0x04bd6269, 0xfde48e17, 0x00d6ef99, 0xffd057bb, 0xffddb374, 0x0042d353, 0xffb8b9f3, 0x003d6aea, 0xffd1d635, 0x001ee3e1, 0xffeda32a, 0x00098070, 0xfffbed0a, 0x000145d7, + 0x787f4134, 0xf3454b6a, 0x051ee498, 0xfda56f9c, 0x0101fe7a, 0xffb2a145, 0xfff2024e, 0x003542e2, 0xffc17d30, 0x0037fe85, 0xffd50530, 0x001d23b9, 0xffee862e, 0x0009192f, 0xfffc1558, 0x0001399e, + 0x7809a65e, 0xf2993d95, 0x057ed264, 0xfd6716f2, 0x012ca389, 0xff952429, 0x000632fa, 0x0027c151, 0xffca39dd, 0x00329483, 0xffd833a0, 0x001b637e, 0xffef695c, 0x0008b1db, 0xfffc3da2, 0x00012d72, + 0x778e4e68, 0xf1f19421, 0x05dd218f, 0xfd298be0, 0x0156d920, 0xff77e470, 0x001a42a4, 0x001a508e, 0xffd2eeb3, 0x002d2db0, 0xffdb6109, 0x0019a373, 0xfff04c8f, 0x00084a86, 0xfffc65e2, 0x00012155, + 0x770d4466, 0xf14e544f, 0x0639c82d, 0xfcecd602, 0x018099b2, 0xff5ae614, 0x002e2e82, 0x000cf281, 0xffdb9a70, 0x0027cada, 0xffde8cf1, 0x0017e3df, 0xfff12fa3, 0x0007e33f, 0xfffc8e11, 0x0001154a, + 0x768693ec, 0xf0af82e4, 0x0694bca0, 0xfcb0fcca, 0x01a9dfcc, 0xff3e2d01, 0x0041f3d2, 0xffffa90e, 0xffe43bd5, 0x00226ccb, 0xffe1b6dd, 0x00162507, 0xfff21275, 0x00077c17, 0xfffcb628, 0x00010952, + 0x75fa4911, 0xf015242b, 0x06edf595, 0xfc76077b, 0x01d2a615, 0xff21bd11, 0x00558fdc, 0xfff27611, 0xffecd1a6, 0x001d144a, 0xffe4de56, 0x0014672d, 0xfff2f4e0, 0x00071520, 0xfffcde20, 0x0000fd6f, + 0x75687068, 0xef7f3bf5, 0x07456a0e, 0xfc3bfd2e, 0x01fae74e, 0xff059a0e, 0x0068fff3, 0xffe55b60, 0xfff55aae, 0x0017c21c, 0xffe802e6, 0x0012aa95, 0xfff3d6c3, 0x0006ae6a, 0xfffd05f3, 0x0000f1a4, + 0x74d11703, 0xeeedcd98, 0x079b1158, 0xfc02e4cc, 0x02229e57, 0xfee9c7af, 0x007c4177, 0xffd85ac9, 0xfffdd5b8, 0x00127704, 0xffeb2416, 0x0010ef82, 0xfff4b7fb, 0x00064804, 0xfffd2d9b, 0x0000e5f3, + 0x74344a70, 0xee60dbee, 0x07eee314, 0xfbcac510, 0x0249c629, 0xfece499d, 0x008f51cf, 0xffcb7615, 0x00064197, 0x000d33c3, 0xffee4174, 0x000f3633, 0xfff59866, 0x0005e1fe, 0xfffd5511, 0x0000da5c, + 0x739218b8, 0xedd86958, 0x0840d732, 0xfb93a486, 0x027059da, 0xfeb3236b, 0x00a22e71, 0xffbeaf06, 0x000e9d1f, 0x0007f915, 0xfff15a8d, 0x000d7eea, 0xfff677e2, 0x00057c68, 0xfffd7c4f, 0x0000cee3, + 0x72ea905a, 0xed5477be, 0x0890e5f7, 0xfb5d898c, 0x029654a0, 0xfe98589b, 0x00b4d4dd, 0xffb20754, 0x0016e72c, 0x0002c7b6, 0xfff46ef1, 0x000bc9e6, 0xfff75650, 0x00051750, 0xfffda350, 0x0000c388, + 0x723dc051, 0xecd5088e, 0x08df07f6, 0xfb287a4d, 0x02bbb1cc, 0xfe7dec9c, 0x00c7429f, 0xffa580b1, 0x001f1e9b, 0xfffda05c, 0xfff77e31, 0x000a1765, 0xfff8338e, 0x0004b2c7, 0xfffdca0d, 0x0000b84d, + 0x718bb80b, 0xec5a1cbc, 0x092b3617, 0xfaf47cc4, 0x02e06ccf, 0xfe63e2cc, 0x00d97550, 0xff991cc9, 0x00274253, 0xfff883be, 0xfffa87df, 0x000867a5, 0xfff90f7c, 0x00044eda, 0xfffdf080, 0x0000ad34, + 0x70d4876b, 0xebe3b4c5, 0x09756994, 0xfac196bb, 0x03048139, 0xfe4a3e70, 0x00eb6a95, 0xff8cdd3c, 0x002f513a, 0xfff3728d, 0xfffd8b92, 0x0006bae1, 0xfff9e9fd, 0x0003eb98, 0xfffe16a6, 0x0000a23f, + 0x70183ec5, 0xeb71d0ab, 0x09bd9bfb, 0xfa8fcdca, 0x0327eab8, 0xfe3102bd, 0x00fd2022, 0xff80c3a4, 0x00374a40, 0xffee6d78, 0x000088df, 0x00051157, 0xfffac2f0, 0x0003890e, 0xfffe3c76, 0x0000976e, + 0x6f56eee1, 0xeb046ffc, 0x0a03c72b, 0xfa5f2755, 0x034aa51b, 0xfe1832d4, 0x010e93b5, 0xff74d194, 0x003f2c57, 0xffe97529, 0x00037f60, 0x00036b3f, 0xfffb9a38, 0x0003274c, 0xfffe61ee, 0x00008cc4, + 0x6e90a8f2, 0xea9b91cc, 0x0a47e559, 0xfa2fa890, 0x036cac52, 0xfdffd1bd, 0x011fc31c, 0xff690894, 0x0046f679, 0xffe48a4a, 0x00066eae, 0x0001c8d2, 0xfffc6fb8, 0x0002c65d, 0xfffe8707, 0x00008241, + 0x6dc57e9b, 0xea3734bb, 0x0a89f10c, 0xfa015679, 0x038dfc6c, 0xfde7e26f, 0x0130ac31, 0xff5d6a24, 0x004ea7a3, 0xffdfad7f, 0x00095666, 0x00002a4a, 0xfffd4352, 0x00026650, 0xfffeabbd, 0x000077e8, + 0x6cf581e8, 0xe9d756f3, 0x0ac9e521, 0xf9d435dc, 0x03ae919a, 0xfdd067ca, 0x01414cdd, 0xff51f7bb, 0x00563edb, 0xffdadf69, 0x000c3627, 0xfffe8fdc, 0xfffe14eb, 0x00020730, 0xfffed00a, 0x00006db9, + 0x6c20c550, 0xe97bf627, 0x0b07bcc6, 0xf9a84b50, 0x03ce682d, 0xfdb96498, 0x0151a317, 0xff46b2c7, 0x005dbb29, 0xffd620a6, 0x000f0d91, 0xfffcf9be, 0xfffee466, 0x0001a90b, 0xfffef3ea, 0x000063b5, + 0x6b475bb0, 0xe9250f99, 0x0b437380, 0xf97d9b37, 0x03ed7c9a, 0xfda2db8c, 0x0161ace5, 0xff3b9cad, 0x00651b9c, 0xffd171d1, 0x0011dc47, 0xfffb6825, 0xffffb1aa, 0x00014bed, 0xffff1759, 0x000059dd, + 0x6a69584a, 0xe8d2a017, 0x0b7d0525, 0xf95429c0, 0x040bcb77, 0xfd8ccf46, 0x01716859, 0xff30b6c8, 0x006c5f4b, 0xffccd380, 0x0014a1ee, 0xfff9db44, 0x00007c9c, 0x0000efe1, 0xffff3a53, 0x00005033, + 0x6986cec4, 0xe884a3fb, 0x0bb46de2, 0xf92bfae4, 0x0429517b, 0xfd77424c, 0x0180d397, 0xff260269, 0x00738551, 0xffc84645, 0x00175e2d, 0xfff8534d, 0x00014521, 0x000094f3, 0xffff5cd2, 0x000046b8, + 0x689fd324, 0xe83b1731, 0x0be9aa34, 0xf9051266, 0x04460b81, 0xfd62370e, 0x018fecd1, 0xff1b80da, 0x007a8cd0, 0xffc3cab1, 0x001a10ad, 0xfff6d070, 0x00020b23, 0x00003b2e, 0xffff7ed3, 0x00003d6c, + 0x67b479cf, 0xe7f5f531, 0x0c1cb6ef, 0xf8df73d6, 0x0461f688, 0xfd4dafe6, 0x019eb246, 0xff113358, 0x008174ef, 0xffbf614e, 0x001cb91a, 0xfff552de, 0x0002ce87, 0xffffe29d, 0xffffa052, 0x00003450, + 0x66c4d787, 0xe7b53908, 0x0c4d913a, 0xf8bb228c, 0x047d0fb1, 0xfd39af17, 0x01ad2249, 0xff071b16, 0x00883cdc, 0xffbb0aa3, 0x001f5723, 0xfff3dac3, 0x00038f37, 0xffff8b4b, 0xffffc14b, 0x00002b66, + 0x65d10168, 0xe778dd50, 0x0c7c368d, 0xf89821ac, 0x0497543f, 0xfd2636ca, 0x01bb3b37, 0xfefd3941, 0x008ee3cd, 0xffb6c735, 0x0021ea76, 0xfff2684e, 0x00044d1b, 0xffff3540, 0xffffe1bc, 0x000022ad, + 0x64d90ce7, 0xe740dc3c, 0x0ca8a4b7, 0xf8767422, 0x04b0c19a, 0xfd134913, 0x01c8fb81, 0xfef38ef6, 0x009568fc, 0xffb29782, 0x002472c8, 0xfff0fba9, 0x0005081f, 0xfffee088, 0x0000019f, 0x00001a28, + 0x63dd0fcd, 0xe70d2f8d, 0x0cd2d9d5, 0xf8561ca7, 0x04c9554e, 0xfd00e7ec, 0x01d661a6, 0xfeea1d4c, 0x009bcbab, 0xffae7c06, 0x0026efcc, 0xffef94fe, 0x0005c02c, 0xfffe8d2c, 0x000020f3, 0x000011d5, + 0x62dd2039, 0xe6ddd09f, 0x0cfad45a, 0xf8371dbb, 0x04e10d0a, 0xfcef153a, 0x01e36c34, 0xfee0e54e, 0x00a20b23, 0xffaa7538, 0x0029613a, 0xffee3477, 0x0006752d, 0xfffe3b35, 0x00003fb3, 0x000009b6, + 0x61d95497, 0xe6b2b862, 0x0d209309, 0xf81979ab, 0x04f7e6a2, 0xfcddd2c7, 0x01f019cb, 0xfed7e7fd, 0x00a826b2, 0xffa6838c, 0x002bc6cd, 0xffecda3b, 0x0007270f, 0xfffdeaaa, 0x00005ddd, 0x000001cc, + 0x60d1c3a6, 0xe68bdf5e, 0x0d4414f9, 0xf7fd328c, 0x050de00d, 0xfccd2246, 0x01fc691b, 0xfecf2650, 0x00ae1dae, 0xffa2a770, 0x002e2040, 0xffeb866f, 0x0007d5bf, 0xfffd9b96, 0x00007b6f, 0xfffffa17, + 0x5fc68470, 0xe6693db5, 0x0d65598f, 0xf7e24a3c, 0x0522f766, 0xfcbd0551, 0x020858e2, 0xfec6a130, 0x00b3ef73, 0xff9ee150, 0x00306d52, 0xffea3939, 0x0008812a, 0xfffd4dff, 0x00009865, 0xfffff297, + 0x5eb7ae46, 0xe64acb24, 0x0d846084, 0xf7c8c267, 0x05372aee, 0xfcad7d6b, 0x0213e7f0, 0xfebe5980, 0x00b99b65, 0xff9b3192, 0x0032adc4, 0xffe8f2bb, 0x0009293e, 0xfffd01ee, 0x0000b4bd, 0xffffeb4c, + 0x5da558c5, 0xe6307f05, 0x0da129df, 0xf7b09c7f, 0x054a7909, 0xfc9e8bfd, 0x021f1526, 0xfeb65015, 0x00bf20ee, 0xff979898, 0x0034e15b, 0xffe7b317, 0x0009cdeb, 0xfffcb769, 0x0000d074, 0xffffe438, + 0x5c8f9bcb, 0xe61a504f, 0x0dbbb5f6, 0xf799d9c4, 0x055ce03f, 0xfc903258, 0x0229df75, 0xfeae85bb, 0x00c47f7f, 0xff9416c1, 0x003707dc, 0xffe67a6f, 0x000a6f20, 0xfffc6e78, 0x0000eb89, 0xffffdd5a, + 0x5b768f7a, 0xe6083599, 0x0dd40571, 0xf7847b3d, 0x056e5f3d, 0xfc8271b4, 0x023445dd, 0xfea6fb32, 0x00c9b691, 0xff90ac66, 0x00392111, 0xffe548e0, 0x000b0cce, 0xfffc2720, 0x000105f9, 0xffffd6b2, + 0x5a5a4c32, 0xe5fa2519, 0x0dea1943, 0xf77081be, 0x057ef4d3, 0xfc754b32, 0x023e4772, 0xfe9fb12e, 0x00cec5a1, 0xff8d59dd, 0x003b2cc5, 0xffe41e88, 0x000ba6e5, 0xfffbe169, 0x00011fc3, 0xffffd041, + 0x593aea93, 0xe5f014aa, 0x0dfdf2ae, 0xf75dede5, 0x058e9ff8, 0xfc68bfd7, 0x0247e354, 0xfe98a85b, 0x00d3ac38, 0xff8a1f77, 0x003d2ac6, 0xffe2fb83, 0x000c3d59, 0xfffb9d59, 0x000138e4, 0xffffca06, + 0x58188376, 0xe5e9f9ca, 0x0e0f9342, 0xf74cc01c, 0x059d5fc5, 0xfc5cd092, 0x025118b8, 0xfe91e159, 0x00d869e1, 0xff86fd81, 0x003f1ae4, 0xffe1dfec, 0x000cd01b, 0xfffb5af3, 0x0001515c, 0xffffc402, + 0x56f32fea, 0xe5e7c99e, 0x0e1efcdb, 0xf73cf898, 0x05ab3377, 0xfc517e38, 0x0259e6e1, 0xfe8b5cba, 0x00dcfe32, 0xff83f443, 0x0040fcf3, 0xffe0cbdc, 0x000d5f1f, 0xfffb1a3f, 0x00016928, 0xffffbe35, + 0x55cb0935, 0xe5e978f0, 0x0e2c319d, 0xf72e9758, 0x05b81a70, 0xfc46c987, 0x02624d23, 0xfe851b09, 0x00e168c5, 0xff810401, 0x0042d0c9, 0xffdfbf6b, 0x000dea5a, 0xfffadb40, 0x00018048, 0xffffb89f, + 0x54a028d0, 0xe5eefc35, 0x0e3733fc, 0xf7219c2a, 0x05c41435, 0xfc3cb323, 0x026a4ae5, 0xfe7f1cc4, 0x00e5a93c, 0xff7e2cfb, 0x0044963d, 0xffdebaaf, 0x000e71c1, 0xfffa9dfa, 0x000196ba, 0xffffb340, + 0x5372a862, 0xe5f8478d, 0x0e4006b2, 0xf71606a6, 0x05cf2070, 0xfc333b97, 0x0271df9c, 0xfe79625e, 0x00e9bf43, 0xff7b6f6c, 0x00464d2b, 0xffddbdbd, 0x000ef549, 0xfffa6273, 0x0001ac7d, 0xffffae17, + 0x5242a1c1, 0xe6054ec6, 0x0e46acc4, 0xf70bd632, 0x05d93eee, 0xfc2a6356, 0x02790ace, 0xfe73ec40, 0x00edaa88, 0xff78cb8c, 0x0047f571, 0xffdcc8a9, 0x000f74e9, 0xfffa28ad, 0x0001c191, 0xffffa924, + 0x51102eec, 0xe616055a, 0x0e4b297c, 0xf7030a01, 0x05e26f9f, 0xfc222abb, 0x027fcc12, 0xfe6ebac6, 0x00f16ac4, 0xff76418b, 0x00498eed, 0xffdbdb84, 0x000ff098, 0xfff9f0ac, 0x0001d5f4, 0xffffa467, + 0x4fdb6a09, 0xe62a5e76, 0x0e4d806f, 0xf6fba113, 0x05eab296, 0xfc1a9208, 0x02862311, 0xfe69ce43, 0x00f4ffb6, 0xff73d199, 0x004b1984, 0xffdaf65e, 0x0010684e, 0xfff9ba73, 0x0001e9a7, 0xffff9fe0, + 0x4ea46d66, 0xe6424cf8, 0x0e4db575, 0xf6f59a36, 0x05f20809, 0xfc139968, 0x028c0f83, 0xfe6526fe, 0x00f86924, 0xff717bdf, 0x004c951b, 0xffda1948, 0x0010dc05, 0xfff98604, 0x0001fca8, 0xffff9b8f, + 0x4d6b536f, 0xe65dc373, 0x0e4bccac, 0xf6f0f407, 0x05f87053, 0xfc0d40ec, 0x0291912f, 0xfe60c533, 0x00fba6da, 0xff6f4083, 0x004e0199, 0xffd9444e, 0x00114bb4, 0xfff95363, 0x00020ef7, 0xffff9773, + 0x4c3036b2, 0xe67cb42f, 0x0e47ca78, 0xf6edacf2, 0x05fdebee, 0xfc07888e, 0x0296a7f0, 0xfe5ca913, 0x00feb8ad, 0xff6d1fa5, 0x004f5ee9, 0xffd8777d, 0x0011b757, 0xfff92290, 0x00022095, 0xffff938c, + 0x4af331d9, 0xe69f112f, 0x0e41b37c, 0xf6ebc332, 0x06027b78, 0xfc027031, 0x029b53af, 0xfe58d2c5, 0x01019e78, 0xff6b1961, 0x0050acf7, 0xffd7b2e0, 0x00121ee9, 0xfff8f38e, 0x00023181, 0xffff8fd9, + 0x49b45fa8, 0xe6c4cc2e, 0x0e398c9f, 0xf6eb34d4, 0x06061fb2, 0xfbfdf79e, 0x029f9466, 0xfe554265, 0x0104581c, 0xff692dd2, 0x0051ebb4, 0xffd6f67f, 0x00128265, 0xfff8c65d, 0x000241bb, 0xffff8c5a, + 0x4873daf7, 0xe6edd6a4, 0x0e2f5b0b, 0xf6ebffb2, 0x0608d97c, 0xfbfa1e88, 0x02a36a1e, 0xfe51f802, 0x0106e583, 0xff675d09, 0x00531b12, 0xffd64264, 0x0012e1c8, 0xfff89b00, 0x00025143, 0xffff890e, + 0x4731beb7, 0xe71a21c7, 0x0e232425, 0xf6ee217b, 0x060aa9da, 0xfbf6e48c, 0x02a6d4f0, 0xfe4ef3a4, 0x0109469d, 0xff65a718, 0x00543b04, 0xffd59695, 0x00133d0e, 0xfff87176, 0x0002601b, 0xffff85f5, + 0x45ee25e7, 0xe7499e8f, 0x0e14ed93, 0xf6f197ad, 0x060b91ee, 0xfbf4492d, 0x02a9d508, 0xfe4c3546, 0x010b7b61, 0xff640c08, 0x00554b83, 0xffd4f316, 0x00139436, 0xfff849c0, 0x00026e41, 0xffff830e, + 0x44a92b96, 0xe77c3db4, 0x0e04bd39, 0xf6f65f9b, 0x060b92ff, 0xfbf24bd9, 0x02ac6a9e, 0xfe49bcd9, 0x010d83cb, 0xff628be3, 0x00564c88, 0xffd457ec, 0x0013e73e, 0xfff823dd, 0x00027bb8, 0xffff805a, + 0x4362eadc, 0xe7b1efb4, 0x0df29936, 0xf6fc766a, 0x060aae6e, 0xfbf0ebe7, 0x02ae95fb, 0xfe478a42, 0x010f5fe2, 0xff6126a9, 0x00573e0f, 0xffd3c519, 0x00143626, 0xfff7ffce, 0x0002887f, 0xffff7dd6, + 0x421b7edf, 0xe7eaa4d4, 0x0dde87e2, 0xf703d912, 0x0608e5c2, 0xfbf02896, 0x02b05779, 0xfe459d5e, 0x01110faf, 0xff5fdc5b, 0x00582016, 0xffd33a9e, 0x001480ec, 0xfff7dd92, 0x00029497, 0xffff7b82, + 0x40d302c5, 0xe8264d21, 0x0dc88fd2, 0xf70c8461, 0x06063a9d, 0xfbf00112, 0x02b1af7f, 0xfe43f5ff, 0x01129344, 0xff5eacf3, 0x0058f29f, 0xffd2b87c, 0x0014c792, 0xfff7bd28, 0x0002a002, 0xffff795f, + 0x3f8991bd, 0xe864d874, 0x0db0b7d1, 0xf71674fa, 0x0602aec3, 0xfbf0746e, 0x02b29e84, 0xfe4293ec, 0x0113eabb, 0xff5d9867, 0x0059b5ad, 0xffd23eaf, 0x00150a19, 0xfff79e8f, 0x0002aac0, 0xffff776a, + 0x3e3f46f2, 0xe8a63671, 0x0d9706e1, 0xf721a756, 0x05fe4414, 0xfbf181a9, 0x02b3250f, 0xfe4176e2, 0x01151632, 0xff5c9eaa, 0x005a6946, 0xffd1cd37, 0x00154883, 0xfff781c5, 0x0002b4d2, 0xffff75a3, + 0x3cf43d8f, 0xe8ea568f, 0x0d7b843b, 0xf72e17c4, 0x05f8fc8f, 0xfbf327ab, 0x02b343b5, 0xfe409e95, 0x011615ce, 0xff5bbfaa, 0x005b0d72, 0xffd1640e, 0x001582d3, 0xfff766c8, 0x0002be3b, 0xffff740a, + 0x3ba890b9, 0xe9312813, 0x0d5e3749, 0xf73bc26b, 0x05f2da52, 0xfbf56549, 0x02b2fb1a, 0xfe400aae, 0x0116e9bc, 0xff5afb53, 0x005ba23b, 0xffd1032f, 0x0015b90b, 0xfff74d97, 0x0002c6fa, 0xffff729e, + 0x3a5c5b8e, 0xe97a9a17, 0x0d3f27ab, 0xf74aa34c, 0x05ebdf97, 0xfbf83941, 0x02b24bf1, 0xfe3fbacd, 0x0117922f, 0xff5a5189, 0x005c27af, 0xffd0aa93, 0x0015eb2f, 0xfff7362f, 0x0002cf12, 0xffff715d, + 0x390fb920, 0xe9c69b8c, 0x0d1e5d32, 0xf75ab63f, 0x05e40eb3, 0xfbfba23f, 0x02b136f9, 0xfe3fae87, 0x01180f5d, 0xff59c230, 0x005c9ddc, 0xffd05a33, 0x00161944, 0xfff7208d, 0x0002d684, 0xffff7047, + 0x37c2c474, 0xea151b3a, 0x0cfbdfdd, 0xf76bf6f7, 0x05db6a19, 0xfbff9ed7, 0x02afbd02, 0xfe3fe569, 0x01186187, 0xff594d27, 0x005d04d4, 0xffd01205, 0x0016434f, 0xfff70caf, 0x0002dd53, 0xffff6f5c, + 0x36759880, 0xea6607c4, 0x0cd7b7dd, 0xf77e6103, 0x05d1f459, 0xfc042d8e, 0x02addee8, 0xfe405ef6, 0x011888f2, 0xff58f249, 0x005d5cab, 0xffcfd1ff, 0x00166956, 0xfff6fa92, 0x0002e37e, 0xffff6e99, + 0x35285026, 0xeab94fa9, 0x0cb1ed8c, 0xf791efcb, 0x05c7b01a, 0xfc094cd2, 0x02ab9d96, 0xfe411aa8, 0x011885e7, 0xff58b16c, 0x005da575, 0xffcf9a15, 0x00168b5e, 0xfff6ea31, 0x0002e90a, 0xffff6dff, + 0x33db0631, 0xeb0ee148, 0x0c8a8973, 0xf7a69e96, 0x05bca021, 0xfc0efafe, 0x02a8fa03, 0xfe4217ef, 0x011858b9, 0xff588a65, 0x005ddf4c, 0xffcf6a3b, 0x0016a96f, 0xfff6db89, 0x0002edf6, 0xffff6d8d, + 0x328dd556, 0xeb66aae0, 0x0c619444, 0xf7bc6889, 0x05b0c74b, 0xfc15365c, 0x02a5f535, 0xfe435633, 0x011801be, 0xff587d03, 0x005e0a48, 0xffcf4262, 0x0016c390, 0xfff6ce97, 0x0002f246, 0xffff6d40, + 0x3140d82e, 0xebc09a94, 0x0c3716da, 0xf7d348a4, 0x05a42890, 0xfc1bfd22, 0x02a2903e, 0xfe44d4d3, 0x01178152, 0xff588913, 0x005e2687, 0xffcf227b, 0x0016d9c9, 0xfff6c356, 0x0002f5fc, 0xffff6d1a, + 0x2ff42933, 0xec1c9e6d, 0x0c0b1a37, 0xf7eb39cc, 0x0596c6ff, 0xfc234d75, 0x029ecc3c, 0xfe469325, 0x0116d7d7, 0xff58ae5d, 0x005e3427, 0xffcf0a77, 0x0016ec22, 0xfff6b9c1, 0x0002f919, 0xffff6d17, + 0x2ea7e2c0, 0xec7aa45b, 0x0bdda783, 0xf80436c0, 0x0588a5bf, 0xfc2b2567, 0x029aaa5a, 0xfe489077, 0x011605b5, 0xff58eca8, 0x005e3347, 0xffcefa44, 0x0016faa5, 0xfff6b1d5, 0x0002fba0, 0xffff6d38, + 0x2d5c1f0e, 0xecda9a39, 0x0baec80a, 0xf81e3a25, 0x0579c812, 0xfc3382fb, 0x02962bd1, 0xfe4acc0e, 0x01150b5a, 0xff5943b4, 0x005e240a, 0xffcef1cf, 0x0017055b, 0xfff6ab8c, 0x0002fd94, 0xffff6d7c, + 0x2c10f82d, 0xed3c6dce, 0x0b7e853c, 0xf8393e81, 0x056a314b, 0xfc3c6420, 0x029151e3, 0xfe4d4526, 0x0113e937, 0xff59b340, 0x005e0694, 0xffcef106, 0x00170c4f, 0xfff6a6e2, 0x0002fef6, 0xffff6de2, + 0x2ac68807, 0xeda00cd1, 0x0b4ce8a8, 0xf8553e3c, 0x0559e4da, 0xfc45c6b6, 0x028c1de0, 0xfe4ffaf6, 0x01129fc5, 0xff5a3b09, 0x005ddb0b, 0xffcef7d4, 0x00170f8a, 0xfff6a3d0, 0x0002ffc9, 0xffff6e67, + 0x297ce85a, 0xee0564e8, 0x0b19fbfe, 0xf87233a4, 0x0548e63f, 0xfc4fa88f, 0x02869122, 0xfe52ecab, 0x01112f81, 0xff5adac6, 0x005da198, 0xffcf0623, 0x00170f18, 0xfff6a252, 0x00030010, 0xffff6f0d, + 0x283432b9, 0xee6c63ad, 0x0ae5c90b, 0xf89018eb, 0x05373912, 0xfc5a076a, 0x0280ad0f, 0xfe561969, 0x010f98eb, 0xff5b922d, 0x005d5a62, 0xffcf1bde, 0x00170b04, 0xfff6a262, 0x0002ffcd, 0xffff6fd1, + 0x26ec8083, 0xeed4f6b0, 0x0ab059bc, 0xf8aee828, 0x0524e100, 0xfc64e0f9, 0x027a7318, 0xfe598050, 0x010ddc8c, 0xff5c60ee, 0x005d0597, 0xffcf38ec, 0x0017035a, 0xfff6a3f9, 0x0002ff03, 0xffff70b2, + 0x25a5eae8, 0xef3f0b78, 0x0a79b814, 0xf8ce9b5d, 0x0511e1c6, 0xfc7032de, 0x0273e4b8, 0xfe5d2075, 0x010bfaee, 0xff5d46bb, 0x005ca363, 0xffcf5d36, 0x0016f828, 0xfff6a713, 0x0002fdb4, 0xffff71b0, + 0x24608ae2, 0xefaa8f87, 0x0a41ee32, 0xf8ef2c71, 0x04fe3f39, 0xfc7bfaad, 0x026d0374, 0xfe60f8ea, 0x0109f4a2, 0xff5e433e, 0x005c33f6, 0xffcf88a2, 0x0016e979, 0xfff6aba9, 0x0002fbe4, 0xffff72c9, + 0x231c7932, 0xf017705a, 0x0a09064e, 0xf9109535, 0x04e9fd3c, 0xfc8835ed, 0x0265d0dd, 0xfe6508b6, 0x0107ca3c, 0xff5f5621, 0x005bb77f, 0xffcfbb17, 0x0016d75b, 0xfff6b1b4, 0x0002f995, 0xffff73fc, + 0x21d9ce63, 0xf0859b6e, 0x09cf0ab4, 0xf932cf65, 0x04d51fc6, 0xfc94e216, 0x025e4e8b, 0xfe694edd, 0x01057c57, 0xff607f0b, 0x005b2e31, 0xffcff478, 0x0016c1dc, 0xfff6b92d, 0x0002f6c9, 0xffff7549, + 0x2098a2bf, 0xf0f4fe3d, 0x099405c6, 0xf955d4a7, 0x04bfaadf, 0xfca1fc96, 0x02567e22, 0xfe6dca58, 0x01030b8e, 0xff61bd9f, 0x005a9840, 0xffd034ac, 0x0016a90a, 0xfff6c20f, 0x0002f385, 0xffff76ae, + 0x1f590e55, 0xf1658649, 0x095801f8, 0xf9799e8f, 0x04a9a29e, 0xfcaf82ca, 0x024e614c, 0xfe727a1f, 0x01007885, 0xff631180, 0x0059f5e1, 0xffd07b95, 0x00168cf2, 0xfff6cc52, 0x0002efca, 0xffff782a, + 0x1e1b28f2, 0xf1d72114, 0x091b09d1, 0xf99e269e, 0x04930b2b, 0xfcbd7206, 0x0245f9bf, 0xfe775d1f, 0x00fdc3e0, 0xff647a4b, 0x0059474a, 0xffd0c915, 0x00166da5, 0xfff6d7f0, 0x0002eb9c, 0xffff79bc, + 0x1cdf0a20, 0xf249bc2c, 0x08dd27e6, 0xf9c36642, 0x047be8bc, 0xfccbc793, 0x023d4937, 0xfe7c7243, 0x00faee49, 0xff65f79e, 0x00588cb4, 0xffd11d0f, 0x00164b32, 0xfff6e4e1, 0x0002e6fe, 0xffff7b63, + 0x1ba4c923, 0xf2bd4523, 0x089e66dd, 0xf9e956da, 0x04643f95, 0xfcda80ad, 0x0234517a, 0xfe81b86d, 0x00f7f86e, 0xff678912, 0x0057c658, 0xffd17764, 0x001625a7, 0xfff6f31d, 0x0002e1f3, 0xffff7d1f, + 0x1a6c7cf9, 0xf331a99b, 0x085ed167, 0xfa0ff1b6, 0x044c1409, 0xfce99a86, 0x022b1455, 0xfe872e7c, 0x00f4e2ff, 0xff692e3f, 0x0056f471, 0xffd1d7f5, 0x0015fd15, 0xfff7029f, 0x0002dc7d, 0xffff7eed, + 0x19363c54, 0xf3a6d741, 0x081e7241, 0xfa373017, 0x04336a75, 0xfcf91246, 0x0221939d, 0xfe8cd349, 0x00f1aeb2, 0xff6ae6ba, 0x0056173b, 0xffd23ea1, 0x0015d18b, 0xfff7135d, 0x0002d6a0, 0xffff80cd, + 0x18021d9d, 0xf41cbbd3, 0x07dd5430, 0xfa5f0b30, 0x041a4744, 0xfd08e50c, 0x0217d12d, 0xfe92a5a7, 0x00ee5c3e, 0xff6cb218, 0x00552ef3, 0xffd2ab47, 0x0015a31b, 0xfff72551, 0x0002d060, 0xffff82bf, + 0x16d036eb, 0xf493451f, 0x079b8203, 0xfa877c29, 0x0400aeec, 0xfd190fed, 0x020dcee8, 0xfe98a466, 0x00eaec5e, 0xff6e8fe9, 0x00543bd8, 0xffd31dc7, 0x001571d5, 0xfff73873, 0x0002c9be, 0xffff84c0, + 0x15a09e09, 0xf50a610a, 0x0759068f, 0xfab07c1d, 0x03e6a5ee, 0xfd298ff6, 0x02038eb7, 0xfe9ece4f, 0x00e75fd1, 0xff707fbd, 0x00533e29, 0xffd395fd, 0x00153dca, 0xfff74cba, 0x0002c2be, 0xffff86d0, + 0x1473686d, 0xf581fd8b, 0x0715ecae, 0xfada0420, 0x03cc30d4, 0xfd3a622b, 0x01f9128a, 0xfea52227, 0x00e3b758, 0xff728121, 0x00523626, 0xffd413c9, 0x0015070b, 0xfff76220, 0x0002bb64, 0xffff88ee, + 0x1348ab3a, 0xf5fa08b5, 0x06d23f3d, 0xfb040d3b, 0x03b15431, 0xfd4b8389, 0x01ee5c55, 0xfeab9eb2, 0x00dff3b7, 0xff7493a2, 0x00512412, 0xffd49705, 0x0014cdab, 0xfff7789c, 0x0002b3b3, 0xffff8b19, + 0x12207b3e, 0xf67270b1, 0x068e091c, 0xfb2e906f, 0x039614a1, 0xfd5cf105, 0x01e36e14, 0xfeb242ac, 0x00dc15b4, 0xff76b6ca, 0x0050082f, 0xffd51f90, 0x001491b9, 0xfff79026, 0x0002abad, 0xffff8d50, + 0x10faecee, 0xf6eb23c6, 0x0649552a, 0xfb5986b6, 0x037a76c7, 0xfd6ea790, 0x01d849c7, 0xfeb90cce, 0x00d81e1a, 0xff78ea20, 0x004ee2c1, 0xffd5ad44, 0x00145349, 0xfff7a8b6, 0x0002a357, 0xffff8f92, + 0x0fd81464, 0xf7641059, 0x06042e45, 0xfb84e906, 0x035e7f4e, 0xfd80a411, 0x01ccf173, 0xfebffbd0, 0x00d40db3, 0xff7b2d2d, 0x004db40c, 0xffd63ffe, 0x0014126c, 0xfff7c245, 0x00029ab2, 0xffff91de, + 0x0eb80562, 0xf7dd24ef, 0x05be9f49, 0xfbb0b04e, 0x034232e6, 0xfd92e36c, 0x01c16720, 0xfec70e64, 0x00cfe54f, 0xff7d7f76, 0x004c7c55, 0xffd6d798, 0x0013cf36, 0xfff7dcc8, 0x000291c3, 0xffff9434, + 0x0d9ad348, 0xf856502d, 0x0578b30e, 0xfbdcd57a, 0x03259644, 0xfda5627e, 0x01b5acdd, 0xfece433a, 0x00cba5bc, 0xff7fe07f, 0x004b3be3, 0xffd773ed, 0x001389b7, 0xfff7f83a, 0x0002888c, 0xffff9691, + 0x0c80911b, 0xf8cf80de, 0x05327467, 0xfc095174, 0x0308ae24, 0xfdb81e22, 0x01a9c4bc, 0xfed598fe, 0x00c74fce, 0xff824fca, 0x0049f2fc, 0xffd814d7, 0x00134204, 0xfff81490, 0x00027f11, 0xffff98f5, + 0x0b69517e, 0xf948a5f0, 0x04ebee1c, 0xfc361d25, 0x02eb7f44, 0xfdcb132d, 0x019db0d0, 0xfedd0e5c, 0x00c2e457, 0xff84ccdb, 0x0048a1e7, 0xffd8ba31, 0x0012f82e, 0xfff831c3, 0x00027555, 0xffff9b60, + 0x0a5526b0, 0xf9c1ae7b, 0x04a52af2, 0xfc633173, 0x02ce0e67, 0xfdde3e6f, 0x01917334, 0xfee4a1fa, 0x00be642f, 0xff875731, 0x004748ed, 0xffd963d4, 0x0012ac48, 0xfff84fcb, 0x00026b5b, 0xffff9dd0, + 0x0944228e, 0xfa3a89be, 0x045e359f, 0xfc908746, 0x02b0604f, 0xfdf19cb9, 0x01850e00, 0xfeec527e, 0x00b9d02b, 0xff89ee4d, 0x0045e856, 0xffda1199, 0x00125e66, 0xfff86e9e, 0x00026126, 0xffffa045, + 0x08365690, 0xfab32723, 0x041718d2, 0xfcbe1789, 0x029279c4, 0xfe052ad4, 0x01788354, 0xfef41e8c, 0x00b52925, 0xff8c91ad, 0x0044806c, 0xffdac35a, 0x00120e9b, 0xfff88e35, 0x000256b9, 0xffffa2be, + 0x072bd3c5, 0xfb2b7641, 0x03cfdf29, 0xfcebdb26, 0x02745f8c, 0xfe18e58c, 0x016bd54f, 0xfefc04c6, 0x00b06ff7, 0xff8f40d0, 0x00431177, 0xffdb78ef, 0x0011bcf9, 0xfff8ae88, 0x00024c18, 0xffffa539, + 0x0624aad6, 0xfba366df, 0x03889336, 0xfd19cb0e, 0x02561670, 0xfe2cc9a7, 0x015f0612, 0xff0403cc, 0x00aba57c, 0xff91fb31, 0x00419bc2, 0xffdc3231, 0x00116994, 0xfff8cf8d, 0x00024146, 0xffffa7b7, + 0x0520ec00, 0xfc1ae8f2, 0x03413f7b, 0xfd47e035, 0x0237a337, 0xfe40d3ed, 0x015217c0, 0xff0c1a3c, 0x00a6ca90, 0xff94c04f, 0x00401f98, 0xffdceef9, 0x00111480, 0xfff8f13c, 0x00023645, 0xffffaa35, + 0x0420a716, 0xfc91eca1, 0x02f9ee68, 0xfd761395, 0x02190aa6, 0xfe550124, 0x01450c7f, 0xff1446b5, 0x00a1e00f, 0xff978fa6, 0x003e9d42, 0xffddaf1e, 0x0010bdcf, 0xfff9138e, 0x00022b19, 0xffffacb4, + 0x0323eb7f, 0xfd086246, 0x02b2aa5c, 0xfda45e2c, 0x01fa5183, 0xfe694e12, 0x0137e672, 0xff1c87d3, 0x009ce6d8, 0xff9a68b0, 0x003d150d, 0xffde727a, 0x00106595, 0xfff93679, 0x00021fc5, 0xffffaf33, +}; + +// cmd-line: fir -l 7 -s 44100 -c 19876 -n 16 -b 9.62 +const int32_t dn_sampler_filter_coefficients[] __attribute__ ((aligned (32))) = { + 0x736144b5, 0x0c333a22, 0xf4fca390, 0x09424904, 0xf8c92a41, 0x052ac04c, 0xfca4fc64, 0x01ed8cc7, 0xff119cc0, 0x0053ba6e, 0xfff9a80d, 0xffeaeaab, 0x001690d9, 0xfff11dcd, 0x000715d9, 0xfffdb4b9, + 0x735ed3aa, 0x0b433de8, 0xf560f0f3, 0x091282c4, 0xf8dd5ccf, 0x0525cb66, 0xfca23e3d, 0x01f33960, 0xff0bc9c2, 0x00586127, 0xfff68603, 0xffecbad5, 0x0015ab8b, 0xfff17c10, 0x0006f71a, 0xfffdbc2f, + 0x735780bb, 0x0a55a98f, 0xf5c5b2a1, 0x08e1ea27, 0xf8f25767, 0x0520366d, 0xfc9ff262, 0x01f89c98, 0xff0620a4, 0x005cf349, 0xfff36c0d, 0xffee8913, 0x0014c5dc, 0xfff1db1a, 0x0006d7d7, 0xfffdc3db, + 0x734b4c77, 0x096a8a51, 0xf62adb7c, 0x08b086aa, 0xf9081629, 0x051a030f, 0xfc9e186a, 0x01fdb637, 0xff00a1d8, 0x00617065, 0xfff05a84, 0xfff0552d, 0x0013dfed, 0xfff23ada, 0x0006b817, 0xfffdcbba, + 0x733a37d2, 0x0881ed1f, 0xf6905e79, 0x087e5fd7, 0xf91e9521, 0x05133308, 0xfc9cafe0, 0x0202860e, 0xfefb4dc7, 0x0065d80c, 0xffed51bc, 0xfff21ee8, 0x0012f9de, 0xfff29b40, 0x000697e0, 0xfffdd3ca, + 0x7324441e, 0x079bdea7, 0xf6f62e9d, 0x084b7d43, 0xf935d048, 0x050bc828, 0xfc9bb83e, 0x02070bf9, 0xfef624d8, 0x006a29d6, 0xffea520a, 0xfff3e60f, 0x001213d0, 0xfff2fc3d, 0x00067739, 0xfffddc07, + 0x7309730f, 0x06b86b52, 0xf75c3eff, 0x0817e68c, 0xf94dc388, 0x0503c44d, 0xfc9b30f3, 0x020b47dd, 0xfef12766, 0x006e655c, 0xffe75bbe, 0xfff5aa69, 0x00112de1, 0xfff35dc1, 0x00065629, 0xfffde470, + 0x72e9c6b8, 0x05d79f40, 0xf7c282cb, 0x07e3a35a, 0xf9666ab7, 0x04fb2969, 0xfc9b195f, 0x020f39ab, 0xfeec55cc, 0x00728a3d, 0xffe46f2a, 0xfff76bc2, 0x00104831, 0xfff3bfbc, 0x000634b6, 0xfffded03, + 0x72c5418e, 0x04f98649, 0xf828ed43, 0x07aebb5d, 0xf97fc19e, 0x04f1f97c, 0xfc9b70d6, 0x0212e15c, 0xfee7b059, 0x0076981a, 0xffe18c9a, 0xfff929e3, 0x000f62de, 0xfff4221f, 0x000612e8, 0xfffdf5bc, + 0x729be665, 0x041e2bfe, 0xf88f71bf, 0x0779364a, 0xf999c3f4, 0x04e83697, 0xfc9c369c, 0x02163ef1, 0xfee33759, 0x007a8e98, 0xffdeb45b, 0xfffae49b, 0x000e7e08, 0xfff484db, 0x0005f0c4, 0xfffdfe9b, + 0x726db871, 0x03459ba4, 0xf8f603ae, 0x07431bdf, 0xf9b46d64, 0x04dde2da, 0xfc9d69eb, 0x02195278, 0xfedeeb11, 0x007e6d61, 0xffdbe6b6, 0xfffc9bb4, 0x000d99cc, 0xfff4e7e1, 0x0005ce51, 0xfffe079b, + 0x723abb44, 0x026fe039, 0xf95c9699, 0x070c73dd, 0xf9cfb988, 0x04d30074, 0xfc9f09ee, 0x021c1c06, 0xfedacbbf, 0x00823422, 0xffd923f4, 0xfffe4efd, 0x000cb647, 0xfff54b20, 0x0005ab95, 0xfffe10bc, + 0x7202f2d3, 0x019d046d, 0xf9c31e22, 0x06d5460b, 0xf9eba3ef, 0x04c791a4, 0xfca115c5, 0x021e9bbb, 0xfed6d99c, 0x0085e28b, 0xffd66c59, 0xfffffe46, 0x000bd397, 0xfff5ae8c, 0x00058898, 0xfffe19fa, + 0x71c6636d, 0x00cd12a4, 0xfa298e07, 0x069d9a31, 0xfa082817, 0x04bb98b5, 0xfca38c83, 0x0220d1bf, 0xfed314da, 0x00897851, 0xffd3c02a, 0x0001a95d, 0x000af1d9, 0xfff61214, 0x0005655e, 0xfffe2354, + 0x718511c2, 0x000014f8, 0xfa8fda21, 0x0665781b, 0xfa254176, 0x04af1804, 0xfca66d2e, 0x0222be45, 0xfecf7da3, 0x008cf52d, 0xffd11fa9, 0x00035015, 0x000a1129, 0xfff675ab, 0x000541f0, 0xfffe2cc8, + 0x713f02e0, 0xff361534, 0xfaf5f669, 0x062ce795, 0xfa42eb75, 0x04a211f8, 0xfca9b6bf, 0x02246187, 0xfecc141d, 0x009058da, 0xffce8b13, 0x0004f23e, 0x000931a3, 0xfff6d942, 0x00051e52, 0xfffe3652, + 0x70f43c32, 0xfe6f1cd7, 0xfb5bd6f4, 0x05f3f06b, 0xfa61216f, 0x04948906, 0xfcad6827, 0x0225bbca, 0xfec8d867, 0x0093a31a, 0xffcc02a8, 0x00068fad, 0x00085362, 0xfff73ccb, 0x0004fa8b, 0xfffe3ff2, + 0x70a4c37f, 0xfdab350f, 0xfbc16ff6, 0x05ba9a6b, 0xfa7fdeba, 0x04867fb3, 0xfcb18047, 0x0226cd5b, 0xfec5ca9a, 0x0096d3af, 0xffc986a1, 0x00082835, 0x00077681, 0xfff7a037, 0x0004d6a1, 0xfffe49a4, + 0x70509eec, 0xfcea66be, 0xfc26b5c5, 0x0580ed5f, 0xfa9f1e9e, 0x0477f88d, 0xfcb5fdf7, 0x02279691, 0xfec2eaca, 0x0099ea62, 0xffc71738, 0x0009bbab, 0x00069b1b, 0xfff8037a, 0x0004b29a, 0xfffe5367, + 0x6ff7d4f8, 0xfc2cba75, 0xfc8b9cda, 0x0546f10f, 0xfabedc5a, 0x0468f62e, 0xfcbae002, 0x022817ca, 0xfec03901, 0x009ce6fe, 0xffc4b4a4, 0x000b49e6, 0x0005c149, 0xfff86686, 0x00048e7c, 0xfffe5d38, + 0x6f9a6c7f, 0xfb723876, 0xfcf019cd, 0x050cad3f, 0xfadf1328, 0x04597b40, 0xfcc0252b, 0x0228516f, 0xfebdb547, 0x009fc954, 0xffc25f1a, 0x000cd2bd, 0x0004e926, 0xfff8c94c, 0x00046a4c, 0xfffe6716, + 0x6f386cb6, 0xfabae8b2, 0xfd54215c, 0x04d229b1, 0xfaffbe36, 0x04498a72, 0xfcc5cc26, 0x022843f0, 0xfebb5f9b, 0x00a29136, 0xffc016cb, 0x000e5609, 0x000412c9, 0xfff92bc0, 0x00044612, 0xfffe70ff, + 0x6ed1dd2e, 0xfa06d2ca, 0xfdb7a869, 0x04976e20, 0xfb20d8ad, 0x04392684, 0xfccbd3a0, 0x0227efc6, 0xfeb937f9, 0x00a53e7b, 0xffbddbe8, 0x000fd3a3, 0x00033e4c, 0xfff98dd6, 0x000421d2, 0xfffe7aef, + 0x6e66c5ce, 0xf955fe0c, 0xfe1aa3fc, 0x045c8240, 0xfb425db0, 0x0428523d, 0xfcd23a3a, 0x02275572, 0xfeb73e54, 0x00a7d0ff, 0xffbbae9f, 0x00114b67, 0x00026bc6, 0xfff9ef80, 0x0003fd92, 0xfffe84e7, + 0x6df72ed9, 0xf8a87178, 0xfe7d0942, 0x04216dc0, 0xfb64485b, 0x0417106e, 0xfcd8fe8b, 0x0226757e, 0xfeb5729b, 0x00aa48a0, 0xffb98f1c, 0x0012bd30, 0x00019b4e, 0xfffa50b1, 0x0003d957, 0xfffe8ee3, + 0x6d8320e6, 0xf7fe33ba, 0xfedecd90, 0x03e63846, 0xfb8693c6, 0x040563f4, 0xfce01f21, 0x0225507c, 0xfeb3d4b7, 0x00aca542, 0xffb77d88, 0x001428db, 0x0000ccfc, 0xfffab15e, 0x0003b527, 0xfffe98e2, + 0x6d0aa4e6, 0xf7574b2b, 0xff3fe663, 0x03aae970, 0xfba93b01, 0x03f34fb2, 0xfce79a7f, 0x0223e706, 0xfeb26489, 0x00aee6ca, 0xffb57a0b, 0x00158e47, 0x000000e6, 0xfffb117a, 0x00039108, 0xfffea2e1, + 0x6c8dc41f, 0xf6b3bdd3, 0xffa04963, 0x036f88d2, 0xfbcc391d, 0x03e0d697, 0xfcef6f20, 0x022239bc, 0xfeb121ee, 0x00b10d23, 0xffb384ca, 0x0016ed53, 0xffff3721, 0xfffb70fa, 0x00036cfe, 0xfffeacdf, + 0x6c0c882a, 0xf6139169, 0xffffec5f, 0x03341df4, 0xfbef8924, 0x03cdfb99, 0xfcf79b75, 0x02204949, 0xfeb00cbf, 0x00b3183c, 0xffb19de7, 0x001845e0, 0xfffe6fc3, 0xfffbcfd2, 0x00034910, 0xfffeb6db, + 0x6b86faf8, 0xf576cb4e, 0x005ec552, 0x02f8b055, 0xfc13261f, 0x03bac1b4, 0xfd001de8, 0x021e165d, 0xfeaf24cc, 0x00b50805, 0xffafc584, 0x001997d0, 0xfffdaadf, 0xfffc2df6, 0x00032541, 0xfffec0d2, + 0x6afd26cb, 0xf4dd7092, 0x00bcca63, 0x02bd4768, 0xfc370b14, 0x03a72bf0, 0xfd08f4d6, 0x021ba1b2, 0xfeae69e1, 0x00b6dc75, 0xffadfbbe, 0x001ae306, 0xfffce88b, 0xfffc8b5c, 0x00030196, 0xfffecac3, + 0x6a6f1638, 0xf44785f1, 0x0119f1e4, 0x0281ea90, 0xfc5b3309, 0x03933d58, 0xfd121e99, 0x0218ec06, 0xfeaddbc4, 0x00b89584, 0xffac40b3, 0x001c2765, 0xfffc28d9, 0xfffce7f8, 0x0002de16, 0xfffed4ab, + 0x69dcd425, 0xf3b50fd6, 0x01763256, 0x0246a125, 0xfc7f9902, 0x037ef900, 0xfd1b9980, 0x0215f621, 0xfead7a37, 0x00ba3330, 0xffaa947c, 0x001d64d5, 0xfffb6bdd, 0xfffd43c1, 0x0002bac4, 0xfffede8a, + 0x69466bc8, 0xf3261255, 0x01d18265, 0x020b726f, 0xfca43803, 0x036a6201, 0xfd2563d3, 0x0212c0d2, 0xfead44f4, 0x00bbb579, 0xffa8f730, 0x001e9b3a, 0xfffab1a8, 0xfffd9eab, 0x000297a5, 0xfffee85e, + 0x68abe8a8, 0xf29a9133, 0x022bd8ee, 0x01d065a8, 0xfcc90b12, 0x03557b7a, 0xfd2f7bd1, 0x020f4cec, 0xfead3bb2, 0x00bd1c63, 0xffa768e6, 0x001fca7d, 0xfff9fa4d, 0xfffdf8ae, 0x000274be, 0xfffef225, + 0x680d5698, 0xf2128fde, 0x02852cfc, 0x019581f9, 0xfcee0d33, 0x03404890, 0xfd39dfb4, 0x020b9b4c, 0xfead5e22, 0x00be67f6, 0xffa5e9b1, 0x0020f288, 0xfff945dc, 0xfffe51be, 0x00025214, 0xfffefbde, + 0x676ac1bb, 0xf18e1174, 0x02dd75ca, 0x015ace79, 0xfd133970, 0x032acc6d, 0xfd448dae, 0x0207acd4, 0xfeadabef, 0x00bf983d, 0xffa479a2, 0x00221344, 0xfff89465, 0xfffea9d2, 0x00022fa9, 0xffff0587, + 0x66c4367d, 0xf10d18bd, 0x0334aac4, 0x0120522f, 0xfd388ad1, 0x03150a3f, 0xfd4f83eb, 0x0203826c, 0xfeae24c1, 0x00c0ad48, 0xffa318c7, 0x00232c9d, 0xfff7e5f9, 0xffff00e1, 0x00020d84, 0xffff0f1f, + 0x6619c197, 0xf08fa82f, 0x038ac385, 0x00e6140f, 0xfd5dfc63, 0x02ff0538, 0xfd5ac08e, 0x01ff1d04, 0xfeaec838, 0x00c1a728, 0xffa1c72f, 0x00243e7f, 0xfff73aa7, 0xffff56e3, 0x0001eba8, 0xffff18a4, + 0x656b700a, 0xf015c1ee, 0x03dfb7dd, 0x00ac1af9, 0xfd838938, 0x02e8c08e, 0xfd6641b8, 0x01fa7d91, 0xfeaf95f2, 0x00c285f4, 0xffa084e3, 0x002548d9, 0xfff6927e, 0xffffabcd, 0x0001ca18, 0xffff2215, + 0x64b94f22, 0xef9f67cb, 0x04337fcb, 0x00726dbb, 0xfda92c63, 0x02d23f7a, 0xfd720581, 0x01f5a50d, 0xfeb08d86, 0x00c349c4, 0xff9f51eb, 0x00264b9a, 0xfff5ed8b, 0xffffff99, 0x0001a8da, 0xffff2b70, + 0x64036c6f, 0xef2c9b43, 0x04861383, 0x0039130c, 0xfdcee0ff, 0x02bb8537, 0xfd7e09fc, 0x01f0947a, 0xfeb1ae87, 0x00c3f2b6, 0xff9e2e50, 0x002746b2, 0xfff54bdc, 0x0000523d, 0x000187f0, 0xffff34b6, + 0x6349d5c9, 0xeebd5d81, 0x04d76b6b, 0x00001191, 0xfdf4a22a, 0x02a49505, 0xfd8a4d37, 0x01eb4cde, 0xfeb2f884, 0x00c480e9, 0xff9d1a14, 0x00283a12, 0xfff4ad7e, 0x0000a3b3, 0x0001675f, 0xffff3de3, + 0x628c994c, 0xee51af5f, 0x0527801d, 0xffc76fd5, 0xfe1a6b08, 0x028d7223, 0xfd96cd3d, 0x01e5cf44, 0xfeb46b07, 0x00c4f480, 0xff9c1539, 0x002925ae, 0xfff4127d, 0x0000f3f1, 0x00014729, 0xffff46f7, + 0x61cbc559, 0xede99165, 0x05764a68, 0xff8f344f, 0xfe4036c5, 0x02761fd3, 0xfda3880f, 0x01e01cbe, 0xfeb60596, 0x00c54da2, 0xff9b1fc1, 0x002a0979, 0xfff37ae4, 0x000142f1, 0x00012754, 0xffff4ff1, + 0x61076890, 0xed8503c7, 0x05c3c34e, 0xff576560, 0xfe660094, 0x025ea157, 0xfdb07bb0, 0x01da3661, 0xfeb7c7b0, 0x00c58c79, 0xff9a39a9, 0x002ae568, 0xfff2e6bf, 0x000190ac, 0x000107e1, 0xffff58d0, + 0x603f91d5, 0xed24066b, 0x060fe408, 0xff20094d, 0xfe8bc3ad, 0x0246f9f3, 0xfdbda61a, 0x01d41d4a, 0xfeb9b0d3, 0x00c5b132, 0xff9962ec, 0x002bb971, 0xfff25619, 0x0001dd1b, 0x0000e8d4, 0xffff6192, + 0x5f745049, 0xecc698e6, 0x065aa604, 0xfee92646, 0xfeb17b53, 0x022f2cea, 0xfdcb0546, 0x01cdd297, 0xfebbc078, 0x00c5bbfc, 0xff989b85, 0x002c858d, 0xfff1c8fa, 0x00022837, 0x0000ca30, 0xffff6a38, + 0x5ea5b34c, 0xec6cba79, 0x06a402e4, 0xfeb2c261, 0xfed722d0, 0x02173d81, 0xfdd89727, 0x01c7576d, 0xfebdf613, 0x00c5ad0a, 0xff97e36c, 0x002d49b4, 0xfff13f6c, 0x000271fa, 0x0000abf8, 0xffff72be, + 0x5dd3ca7a, 0xec166a19, 0x06ebf483, 0xfe7ce399, 0xfefcb57a, 0x01ff2ef9, 0xfde659af, 0x01c0acf5, 0xfec05114, 0x00c58494, 0xff973a96, 0x002e05df, 0xfff0b977, 0x0002ba5f, 0x00008e30, 0xffff7b26, + 0x5cfea5aa, 0xebc3a669, 0x073274f1, 0xfe478fd2, 0xff222eac, 0x01e70494, 0xfdf44acc, 0x01b9d45b, 0xfec2d0e8, 0x00c542d1, 0xff96a0f8, 0x002eba0a, 0xfff03724, 0x0003015f, 0x000070d9, 0xffff836d, + 0x5c2654ed, 0xeb746dbe, 0x07777e74, 0xfe12ccd1, 0xff4789d1, 0x01cec194, 0xfe026869, 0x01b2ced1, 0xfec574f9, 0x00c4e7fe, 0xff961684, 0x002f6630, 0xffefb87a, 0x000346f6, 0x000053f7, 0xffff8b93, + 0x5b4ae88d, 0xeb28be1f, 0x07bb0b8b, 0xfddea042, 0xff6cc25a, 0x01b66936, 0xfe10b06f, 0x01ab9d8b, 0xfec83caa, 0x00c47459, 0xff959b29, 0x00300a4f, 0xffef3d7f, 0x00038b1d, 0x0000378c, 0xffff9398, + 0x5a6c7108, 0xeae09544, 0x07fd16eb, 0xfdab0fb6, 0xff91d3c6, 0x019dfeb6, 0xfe1f20c5, 0x01a441c2, 0xfecb275e, 0x00c3e824, 0xff952ed7, 0x0030a665, 0xffeec63a, 0x0003cdd1, 0x00001b9a, 0xffff9b7a, + 0x598aff13, 0xea9bf097, 0x083d9b81, 0xfd7820a0, 0xffb6b99f, 0x0185854f, 0xfe2db74f, 0x019cbcb1, 0xfece3472, 0x00c343a4, 0xff94d178, 0x00313a72, 0xffee52b1, 0x00040f0d, 0x00000024, 0xffffa339, + 0x58a6a397, 0xea5acd38, 0x087c9471, 0xfd45d856, 0xffdb6f7c, 0x016d0037, 0xfe3c71f1, 0x01950f98, 0xfed16342, 0x00c2871f, 0xff9482f8, 0x0031c677, 0xffede2e7, 0x00044ecb, 0xffffe52d, 0xffffaad3, + 0x57bf6fae, 0xea1d27f7, 0x08b9fd18, 0xfd143c12, 0xfffff100, 0x015472a1, 0xfe4b4e8c, 0x018d3bb8, 0xfed4b325, 0x00c1b2e0, 0xff944340, 0x00324a74, 0xffed76e3, 0x00048d0a, 0xffffcab5, 0xffffb249, + 0x56d574a2, 0xe9e2fd5b, 0x08f5d10a, 0xfce350f0, 0x002439db, 0x013bdfbc, 0xfe5a4b03, 0x01854258, 0xfed82370, 0x00c0c731, 0xff941236, 0x0032c66e, 0xffed0ea7, 0x0004c9c4, 0xffffb0bf, 0xffffb99a, + 0x55e8c3ee, 0xe9ac49a0, 0x09300c14, 0xfcb31bec, 0x004845cc, 0x01234ab4, 0xfe696534, 0x017d24bf, 0xfedbb373, 0x00bfc463, 0xff93efbf, 0x00333a67, 0xffecaa36, 0x000504f6, 0xffff974d, 0xffffc0c5, + 0x54f96f37, 0xe97908b8, 0x0968aa3b, 0xfc83a1e5, 0x006c10a0, 0x010ab6b0, 0xfe789b01, 0x0174e437, 0xfedf627d, 0x00beaac6, 0xff93dbc0, 0x0033a665, 0xffec4994, 0x00053e9e, 0xffff7e61, 0xffffc7ca, + 0x54078851, 0xe9493649, 0x099fa7bb, 0xfc54e79a, 0x008f9631, 0x00f226d0, 0xfe87ea47, 0x016c820d, 0xfee32fdb, 0x00bd7aae, 0xff93d618, 0x00340a6d, 0xffebecc2, 0x000576b8, 0xffff65fc, 0xffffcea8, + 0x53132138, 0xe91ccdb5, 0x09d5010b, 0xfc26f1ad, 0x00b2d26b, 0x00d99e31, 0xfe9750e8, 0x0163ff90, 0xfee71ad4, 0x00bc3470, 0xff93deaa, 0x00346687, 0xffeb93c3, 0x0005ad41, 0xffff4e20, 0xffffd55f, + 0x521c4c10, 0xe8f3ca12, 0x0a08b2d9, 0xfbf9c49d, 0x00d5c147, 0x00c11feb, 0xfea6ccc3, 0x015b5e11, 0xfeeb22af, 0x00bad866, 0xff93f552, 0x0034babb, 0xffeb3e96, 0x0005e238, 0xffff36ce, 0xffffdbee, + 0x51231b26, 0xe8ce2631, 0x0a3aba09, 0xfbcd64ca, 0x00f85ecf, 0x00a8af0c, 0xfeb65bb9, 0x01529ee3, 0xfeef46b0, 0x00b966e9, 0xff9419ef, 0x00350711, 0xffeaed3c, 0x00061599, 0xffff2007, 0xffffe255, + 0x5027a0e9, 0xe8abdc9d, 0x0a6b13bc, 0xfba1d673, 0x011aa71d, 0x00904ea0, 0xfec5fbac, 0x0149c35a, 0xfef3861a, 0x00b7e055, 0xff944c5a, 0x00354b94, 0xffea9fb6, 0x00064764, 0xffff09ce, 0xffffe894, + 0x4f29efed, 0xe88ce79a, 0x0a99bd47, 0xfb771db9, 0x013c965b, 0x007801aa, 0xfed5aa7e, 0x0140cccb, 0xfef7e02a, 0x00b6450a, 0xff948c6e, 0x0035884f, 0xffea5602, 0x00067797, 0xfffef421, 0xffffeeaa, + 0x4e2a1ae8, 0xe871412a, 0x0ac6b43a, 0xfb4d3e97, 0x015e28c7, 0x005fcb26, 0xfee56614, 0x0137bc8f, 0xfefc541e, 0x00b49568, 0xff94da03, 0x0035bd4e, 0xffea1020, 0x0006a630, 0xfffedf04, 0xfffff498, + 0x4d2834b0, 0xe858e30a, 0x0af1f65d, 0xfb243cea, 0x017f5aad, 0x0047ae09, 0xfef52c54, 0x012e93fc, 0xff00e133, 0x00b2d1d1, 0xff9534f0, 0x0035ea9d, 0xffe9ce0d, 0x0006d32f, 0xfffeca76, 0xfffffa5d, + 0x4c245038, 0xe843c6b5, 0x0b1b81ad, 0xfafc1c6e, 0x01a0286c, 0x002fad3f, 0xff04fb25, 0x0125546c, 0xff0586a0, 0x00b0faaa, 0xff959d0a, 0x0036104b, 0xffe98fc8, 0x0006fe92, 0xfffeb678, 0xfffffff8, + 0x4b1e8091, 0xe831e563, 0x0b435462, 0xfad4e0b9, 0x01c08e78, 0x0017cbae, 0xff14d073, 0x011bff38, 0xff0a439e, 0x00af1059, 0xff961224, 0x00362e66, 0xffe9554c, 0x00072859, 0xfffea30b, 0x0000056a, + 0x4a16d8e5, 0xe823380d, 0x0b696ceb, 0xfaae8d43, 0x01e08952, 0x00000c33, 0xff24aa2a, 0x011295bb, 0xff0f1762, 0x00ad1346, 0xff969412, 0x003644fd, 0xffe91e99, 0x00075084, 0xfffe9030, 0x00000ab3, + 0x490d6c79, 0xe817b76c, 0x0b8dc9ed, 0xfa89255f, 0x02001593, 0xffe871a0, 0xff348639, 0x0109194f, 0xff140121, 0x00ab03da, 0xff9722a5, 0x00365422, 0xffe8eba8, 0x00077712, 0xfffe7de7, 0x00000fd2, + 0x48024ea7, 0xe80f5bfb, 0x0bb06a47, 0xfa64ac3f, 0x021f2fe5, 0xffd0fec1, 0xff446293, 0x00ff8b4f, 0xff19000e, 0x00a8e282, 0xff97bdac, 0x00365be6, 0xffe8bc77, 0x00079c04, 0xfffe6c2f, 0x000014c8, + 0x46f592e2, 0xe80a1df5, 0x0bd14d0b, 0xfa4124f2, 0x023dd505, 0xffb9b656, 0xff543d2e, 0x00f5ed15, 0xff1e135b, 0x00a6afa8, 0xff9864f6, 0x00365c5b, 0xffe89101, 0x0007bf5b, 0xfffe5b0b, 0x00001994, + 0x45e74cad, 0xe807f55b, 0x0bf07186, 0xfa1e9262, 0x025c01c5, 0xffa29b18, 0xff641402, 0x00ec3ffc, 0xff233a39, 0x00a46bbc, 0xff991851, 0x00365594, 0xffe8693f, 0x0007e116, 0xfffe4a79, 0x00001e37, + 0x44d78fa0, 0xe808d9f1, 0x0c0dd738, 0xf9fcf758, 0x0279b30b, 0xff8bafb3, 0xff73e50e, 0x00e2855d, 0xff2873d6, 0x00a2172d, 0xff99d789, 0x003647a5, 0xffe8452d, 0x00080137, 0xfffe3a79, 0x000022b1, + 0x43c66f62, 0xe80cc342, 0x0c297dd9, 0xf9dc567b, 0x0296e5d0, 0xff74f6cc, 0xff83ae52, 0x00d8be92, 0xff2dbf61, 0x009fb26c, 0xff9aa268, 0x003632a2, 0xffe824c5, 0x00081fbf, 0xfffe2b0d, 0x00002701, + 0x42b3ffa9, 0xe813a89f, 0x0c436557, 0xf9bcb24a, 0x02b39724, 0xff5e72fb, 0xff936dd2, 0x00ceecf5, 0xff331c08, 0x009d3deb, 0xff9b78ba, 0x003616a2, 0xffe807ff, 0x00083cb0, 0xfffe1c32, 0x00002b28, + 0x41a05437, 0xe81d8122, 0x0c5b8dd4, 0xf99e0d26, 0x02cfc429, 0xff4826cf, 0xffa3219a, 0x00c511dc, 0xff3888f8, 0x009aba1d, 0xff9c5a47, 0x0035f3b9, 0xffe7eed5, 0x0008580a, 0xfffe0dea, 0x00002f26, + 0x408b80d9, 0xe82a43ac, 0x0c71f7a9, 0xf980694a, 0x02eb6a18, 0xff3214c9, 0xffb2c7b6, 0x00bb2e9f, 0xff3e055d, 0x00982778, 0xff9d46d6, 0x0035ca00, 0xffe7d93f, 0x000871cf, 0xfffe0034, 0x000032fb, + 0x3f759967, 0xe839e6e9, 0x0c86a361, 0xf963c8cc, 0x03068640, 0xff1c3f63, 0xffc25e3b, 0x00b14493, 0xff439064, 0x0095866f, 0xff9e3e30, 0x0035998d, 0xffe7c735, 0x00088a02, 0xfffdf310, 0x000036a8, + 0x3e5eb1bd, 0xe84c6152, 0x0c9991be, 0xf9482da0, 0x03211603, 0xff06a907, 0xffd1e340, 0x00a7550c, 0xff492937, 0x0092d77b, 0xff9f4019, 0x00356279, 0xffe7b8af, 0x0008a0a5, 0xfffde67c, 0x00003a2d, + 0x3d46ddc1, 0xe861a92b, 0x0caac3b5, 0xf92d9997, 0x033b16dc, 0xfef15417, 0xffe154e3, 0x009d615d, 0xff4ecf02, 0x00901b11, 0xffa04c57, 0x003524dd, 0xffe7ada5, 0x0008b5ba, 0xfffdda79, 0x00003d89, + 0x3c2e315a, 0xe879b487, 0x0cba3a6d, 0xf9140e5e, 0x03548659, 0xfedc42e7, 0xfff0b148, 0x00936ad6, 0xff5480f0, 0x008d51ab, 0xffa162ae, 0x0034e0d3, 0xffe7a60d, 0x0008c944, 0xfffdcf05, 0x000040be, + 0x3b14c072, 0xe8947947, 0x0cc7f742, 0xf8fb8d7d, 0x036d621f, 0xfec777be, 0xfffff697, 0x008972c7, 0xff5a3e2c, 0x008a7bc1, 0xffa282e1, 0x00349674, 0xffe7a1de, 0x0008db46, 0xfffdc421, 0x000043cc, + 0x39fa9ef3, 0xe8b1ed1c, 0x0cd3fbc0, 0xf8e4185a, 0x0385a7eb, 0xfeb2f4d9, 0x000f22fe, 0x007f7a7c, 0xff6005e1, 0x008799cd, 0xffa3acb4, 0x003445dc, 0xffe7a10d, 0x0008ebc1, 0xfffdb9cb, 0x000046b2, + 0x38dfe0c6, 0xe8d2058b, 0x0cde49a8, 0xf8cdb036, 0x039d558e, 0xfe9ebc66, 0x001e34b4, 0x00758341, 0xff65d73a, 0x0084ac48, 0xffa4dfe8, 0x0033ef25, 0xffe7a391, 0x0008fabb, 0xfffdb002, 0x00004972, + 0x37c499d0, 0xe8f4b7e9, 0x0ce6e2ea, 0xf8b85631, 0x03b468f1, 0xfe8ad087, 0x002d29f3, 0x006b8e5c, 0xff6bb163, 0x0081b3af, 0xffa61c3e, 0x0033926d, 0xffe7a95f, 0x00090836, 0xfffda6c5, 0x00004c0b, + 0x36a8ddf3, 0xe919f961, 0x0cedc9a7, 0xf8a40b44, 0x03cae014, 0xfe773351, 0x003c00fd, 0x00619d15, 0xff719388, 0x007eb07b, 0xffa76176, 0x00332fcf, 0xffe7b26c, 0x00091435, 0xfffd9e13, 0x00004e7f, + 0x358cc109, 0xe941bef3, 0x0cf30031, 0xf890d048, 0x03e0b90d, 0xfe63e6cb, 0x004ab81b, 0x0057b0ae, 0xff777cd6, 0x007ba32a, 0xffa8af51, 0x0032c769, 0xffe7bead, 0x00091ebd, 0xfffd95eb, 0x000050cd, + 0x347056e3, 0xe96bfd76, 0x0cf6890a, 0xf87ea5f1, 0x03f5f20a, 0xfe50ecf0, 0x00594d9d, 0x004dca68, 0xff7d6c79, 0x00788c36, 0xffaa058d, 0x00325958, 0xffe7ce16, 0x000927d1, 0xfffd8e4d, 0x000052f7, + 0x3353b349, 0xe998a999, 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0xffdd3325, 0x00438b7e, 0xffc3084f, 0x0028da1a, 0xffea48be, 0x00090236, 0xfffd5842, 0x000063c0, + 0x21a8ff7e, 0xed878bf0, 0x0c3e9db5, 0xf7ece68c, 0x04f9edda, 0xfd462df6, 0x01374eda, 0xffada547, 0xffe3149e, 0x00401af9, 0xffc4c4da, 0x00281edd, 0xffea85dc, 0x0008f57a, 0xfffd58c5, 0x000063d0, + 0x209416f2, 0xedd63ee5, 0x0c26611f, 0xf7edc7af, 0x050316e0, 0xfd3a16c0, 0x014288e0, 0xffa4f383, 0xffe8ee72, 0x003ca97b, 0xffc68502, 0x00276046, 0xffeac525, 0x0008e7a7, 0xfffd59b2, 0x000063c6, + 0x1f804ab0, 0xee267f35, 0x0c0cc646, 0xf7efadbd, 0x050b8e8a, 0xfd2e6d0d, 0x014d871b, 0xff9c5cdc, 0xffeebfec, 0x0039377a, 0xffc84881, 0x00269e7a, 0xffeb068a, 0x0008d8c4, 0xfffd5b05, 0x000063a3, + 0x1e6dac83, 0xee783f9e, 0x0bf1d2d0, 0xf7f296d7, 0x051354d5, 0xfd2331b0, 0x01584883, 0xff93e241, 0xfff48859, 0x0035c56c, 0xffca0f14, 0x0025d99b, 0xffeb49fc, 0x0008c8d7, 0xfffd5cbe, 0x00006368, + 0x1d5c4e09, 0xeecb72d1, 0x0bd58c81, 0xf7f68103, 0x051a69d4, 0xfd18656f, 0x0162cc19, 0xff8b8498, 0xfffa470a, 0x003253c6, 0xffcbd876, 0x002511cd, 0xffeb8f6a, 0x0008b7e7, 0xfffd5ed8, 0x00006316, + 0x1c4c40b6, 0xef200b76, 0x0bb7f940, 0xf7fb6a29, 0x0520cdb1, 0xfd0e08fb, 0x016d10e9, 0xff8344c4, 0xfffffb51, 0x002ee2fa, 0xffcda463, 0x00244733, 0xffebd6c4, 0x0008a5fa, 0xfffd6154, 0x000062ad, + 0x1b3d95d1, 0xef75fc2b, 0x0b991f0f, 0xf8015015, 0x052680ae, 0xfd041cfa, 0x01771608, 0xff7b23a1, 0x0005a483, 0x002b737b, 0xffcf7299, 0x002379ef, 0xffec1ffa, 0x00089316, 0xfffd642d, 0x0000622e, + 0x1a305e70, 0xefcd3787, 0x0b79040c, 0xf8083077, 0x052b8320, 0xfcfaa200, 0x0180da94, 0xff732209, 0x000b41fa, 0x002805ba, 0xffd142d3, 0x0022aa26, 0xffec6afc, 0x00087f43, 0xfffd6762, 0x0000619a, + 0x1924ab7b, 0xf025b01a, 0x0b57ae75, 0xf81008e2, 0x052fd573, 0xfcf19894, 0x018a5db5, 0xff6b40cb, 0x0010d30e, 0x00249a28, 0xffd314cf, 0x0021d7fa, 0xffecb7b9, 0x00086a86, 0xfffd6af1, 0x000060f1, + 0x181a8da5, 0xf07f586e, 0x0b3524a0, 0xf818d6cf, 0x0533782a, 0xfce9012c, 0x01939e9e, 0xff6380b5, 0x00165720, 0x00213134, 0xffd4e84a, 0x00210390, 0xffed0621, 0x000854e6, 0xfffd6ed6, 0x00006035, + 0x17121573, 0xf0da230b, 0x0b116cff, 0xf822979b, 0x05366bdc, 0xfce0dc2f, 0x019c9c8b, 0xff5be28d, 0x001bcd8e, 0x001dcb4a, 0xffd6bd01, 0x00202d09, 0xffed5624, 0x00083e6a, 0xfffd7310, 0x00005f66, + 0x160b5331, 0xf1360276, 0x0aec8e1c, 0xf82d488c, 0x0538b136, 0xfcd929f4, 0x01a556c1, 0xff546713, 0x002135bd, 0x001a68d8, 0xffd892b4, 0x001f5489, 0xffeda7b1, 0x00082718, 0xfffd779d, 0x00005e84, + 0x150656f8, 0xf192e932, 0x0ac68e9b, 0xf838e6c9, 0x053a48fa, 0xfcd1eac3, 0x01adcc91, 0xff4d0f02, 0x00268f13, 0x00170a47, 0xffda6921, 0x001e7a33, 0xffedfab8, 0x00080ef7, 0xfffd7c7a, 0x00005d92, + 0x140330a9, 0xf1f0c9c5, 0x0a9f7537, 0xf8456f65, 0x053b3400, 0xfccb1ed7, 0x01b5fd54, 0xff45db10, 0x002bd8fa, 0x0013b003, 0xffdc4007, 0x001d9e2a, 0xffee4f29, 0x0007f60f, 0xfffd81a4, 0x00005c8e, + 0x1301efed, 0xf24f96b5, 0x0a7748c0, 0xf852df56, 0x053b7332, 0xfcc4c658, 0x01bde86f, 0xff3ecbea, 0x003112e0, 0x00105a72, 0xffde1726, 0x001cc091, 0xffeea4f2, 0x0007dc65, 0xfffd8719, 0x00005b7b, + 0x1202a434, 0xf2af428c, 0x0a4e101f, 0xf861337c, 0x053b0791, 0xfcbee162, 0x01c58d50, 0xff37e23b, 0x00363c35, 0x000d09fc, 0xffdfee3f, 0x001be18a, 0xffeefc04, 0x0007c201, 0xfffd8cd7, 0x00005a58, + 0x11055cb4, 0xf30fbfd7, 0x0a23d24e, 0xf870689f, 0x0539f231, 0xfcb97001, 0x01cceb6e, 0xff311ea4, 0x003b546b, 0x0009bf05, 0xffe1c511, 0x001b0138, 0xffef544e, 0x0007a6e9, 0xfffd92db, 0x00005927, + 0x100a2864, 0xf371012c, 0x09f8965d, 0xf8807b70, 0x0538343a, 0xfcb47232, 0x01d4024c, 0xff2a81c4, 0x00405afa, 0x000679f2, 0xffe39b60, 0x001a1fbc, 0xffefadc0, 0x00078b24, 0xfffd9923, 0x000057e9, + 0x0f111603, 0xf3d2f926, 0x09cc636e, 0xf8916889, 0x0535cee9, 0xfcafe7e2, 0x01dad175, 0xff240c2f, 0x00454f5d, 0x00033b23, 0xffe570ed, 0x00193d3a, 0xfff00849, 0x00076eba, 0xfffd9fac, 0x0000569d, + 0x0e1a340d, 0xf4359a6a, 0x099f40b5, 0xf8a32c6e, 0x0532c38c, 0xfcabd0f2, 0x01e15880, 0xff1dbe77, 0x004a310f, 0x000002f9, 0xffe7457c, 0x001859d2, 0xfff063d9, 0x000751b0, 0xfffda675, 0x00005545, + 0x0d2590c3, 0xf498d7a5, 0x09713575, 0xf8b5c38d, 0x052f1386, 0xfca82d32, 0x01e7970e, 0xff179926, 0x004eff94, 0xfffcd1d3, 0xffe918ce, 0x001775a7, 0xfff0c060, 0x0007340d, 0xfffdad79, 0x000053e2, + 0x0c333a22, 0xf4fca390, 0x09424904, 0xf8c92a41, 0x052ac04c, 0xfca4fc64, 0x01ed8cc7, 0xff119cc0, 0x0053ba6e, 0xfff9a80d, 0xffeaeaab, 0x001690d9, 0xfff11dcd, 0x000715d9, 0xfffdb4b9, 0x00005274, +}; +} diff --git a/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h b/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h deleted file mode 100644 index 8c6a899..0000000 --- a/services/audioflinger/audio-resampler/resampler_filter_coefficients_10042011.h +++ /dev/null @@ -1,2071 +0,0 @@ -/* - * Copyright (C) 2012 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ -#include <stdlib.h> - -namespace android { - -const int32_t resampler_filter_coefficients_10042011[] = { -2075076504, -2074870219, -2074269557, -2073262841, -2071862786, -2070051926, -2067849110, -2065243563, -2062248465, -2058846262, -2055055548, -2050866069, -2046291635, -2041315273, -2035955897, -2030204167, -2024074532, -2017550518, -2010651175, -2003368165, -1995716923, -1987682243, -1979283938, -1970514752, -1961390076, -1951894181, -1942045775, -1931837943, -1921286728, -1910377736, -1899130344, -1887538902, -1875619770, -1863358864, -1850775104, -1837863721, -1824641940, -1811097380, -1797249776, -1783095528, -1768651291, -1753903742, -1738870726, -1723548746, -1707955054, -1692078274, -1675937190, -1659529795, -1642873313, -1625956474, -1608797063, -1591393772, -1573764495, -1555899971, -1537818785, -1519520784, -1501022796, -1482314240, -1463411160, -1444313227, -1425037741, -1405576492, -1385946622, -1366149382, -1346201706, -1326095287, -1305845828, -1285455045, -1264940308, -1244295464, -1223536951, -1202667530, -1181703085, -1160635857, -1139479271, -1118235636, -1096921134, -1075530585, -1054078577, -1032568949, -1011017224, -989417949, -967783970, -946119336, -924439752, -902741900, -881039249, -859336703, -837648175, -815968557, -794308053, -772670861, -751070988, -729505896, -707986911, -686519594, -665117063, -643776375, -622506966, -601314359, -580211607, -559197735, -538282662, -517472516, -496778402, -476197491, -455736426, -435400430, -415200497, -395136166, -375215078, -355443561, -335831583, -316378199, -297089183, -277970603, -259032151, -240274557, -221703918, -203326638, -185150446, -167174280, -149401179, -131836509, -114487708, -97355714, -80444376, -63759844, -47308498, -31090767, -15108809, --631654, --16124677, --31368536, --46360908, --61095999, --75569692, --89781876, --103732869, --117417959, --130833461, --143977734, --156850513, --169446641, --181763420, --193799687, --205556559, --217029409, --228216068, --239114500, --249725794, --260045485, --270072966, --279807538, --289252364, --298404017, --307262396, --315825804, --324097052, --332072584, --339753137, --347137398, --354229167, --361025450, --367527527, --373733643, --379647676, --385266970, --390594013, --395628043, --400374357, --404831217, --409001628, --412884357, --416484569, --419800667, --422836272, --425590412, --428068799, --430270358, --432199204, --433854305, --435241518, --436360214, --437215297, --437806349, --438139967, --438216244, --438040549, --437612462, --436938620, --436019455, --434860689, --433462013, --431830172, --429966019, --427875663, --425559045, --423023081, --420269108, --417303590, --414126712, --410745497, --407161766, --403382348, --399407797, --395245305, --390897144, --386370224, --381665062, --376788564, --371743279, --366536360, --361168782, --355647611, --349975843, --344160581, --338202705, --332108851, --325882239, --319530194, --313054213, --306461164, --299754744, --292942106, --286024571, --279008412, --271897436, --264698872, --257414606, --250051032, --242612307, --235105276, --227531464, --219896432, --212204319, --204462036, --196671840, --188839540, --180969700, --173068854, --165138974, --157185076, --149211664, --141225186, --133228184, --125225726, --117222535, --109224429, --101233397, --93253442, --85289050, --77345947, --69426846, --61535955, --53678071, --45858404, --38079310, --30344121, --22657427, --15024209, --7447312, -69952, -7522929, -14907403, -22221174, -29462063, -36625813, -43708432, -50707112, -57619541, -64441301, -71168979, -77800149, -84333341, -90764454, -97090405, -103308422, -109417126, -115412500, -121292248, -127054242, -132698149, -138220442, -143619140, -148891701, -154037743, -159053739, -163938275, -168689170, -173306749, -177787867, -182131495, -186335286, -190399716, -194321853, -198101380, -201736519, -205228586, -208575166, -211776310, -214830056, -217737773, -220497194, -223108861, -225571114, -227885824, -230051117, -232067922, -233934597, -235653195, -237222146, -238642925, -239914316, -241038916, -242015623, -242846277, -243529706, -244068632, -244462216, -244712654, -244818983, -244784181, -244607765, -244292268, -243836913, -243244874, -242516010, -241653201, -240655918, -239527557, -238268354, -236881508, -235366743, -233727638, -231964766, -230081521, -228077731, -225956985, -223720128, -221370808, -218909157, -216338938, -213661333, -210880096, -207995414, -205010946, -201928111, -198750891, -195479864, -192118892, -188669741, -185136417, -181519490, -177822602, -174047687, -170198894, -166277172, -162286302, -158228508, -154107834, -149925117, -145683777, -141386129, -137036351, -132635752, -128187958, -123695608, -119162747, -114590586, -109982363, -105340765, -100669861, -95971242, -91248228, -86503703, -81741443, -76962776, -72170457, -67367307, -62557108, -57741653, -52923856, -48106777, -43293939, -38486954, -33688240, -28900789, -24128036, -19371931, -14634917, -9920080, -5230446, -567618, --4066612, --8669336, --13237622, --17769457, --22262939, --26715017, --31123091, --35485370, --39800471, --44065504, --48278050, --52436071, --56538223, --60581634, --64564356, --68484740, --72342096, --76133828, --79858170, --83513191, --87098163, --90610469, --94048704, --97411174, --100697628, --103905676, --107034151, --110081227, --113046738, --115928406, --118725517, --121436605, --124062060, --126599916, --129049689, --131409767, --133680567, --135860199, --137948517, --139944122, --141847799, --143657917, --145374591, --146996418, --148524310, --149956828, --151294475, --152536152, --153683178, --154734431, --155690674, --156550825, --157316298, --157986149, --158561410, --159041176, --159427086, --159718451, --159916548, --160020582, --160032332, --159951337, --159779143, --159515147, --159161329, --158717499, --158185432, --157564678, --156857347, --156063474, --155185008, --154221601, --153175426, --152046727, --150837654, --149548057, --148180246, --146734704, --145213706, --143617181, --141947434, --140205146, --138392776, --136510514, --134560819, --132544622, --130464445, --128320507, --126115178, --123849532, --121526221, --119145720, --116710510, --114221879, --111682456, --109092675, --106454825, --103770291, --101041818, --98270160, --95457755, --92606223, --89718259, --86794577, --83837395, --80848397, --77830331, --74784179, --71712237, --68616354, --65499123, --62361390, --59205117, --56032152, --52845129, --49645224, --46434530, --43215084, --39989380, --36758496, --33524217, --30288565, --27054001, --23821849, --20593926, --17372347, --14159320, --10955954, --7763647, --4584416, --1420431, -1726908, -4856115, -7965053, -11051765, -14114989, -17153587, -20165512, -23148908, -26102328, -29024647, -31913805, -34768253, -37586799, -40368755, -43112232, -45815788, -48478013, -51098183, -53674369, -56205373, -58689945, -61127693, -63516834, -65856322, -68144797, -70381920, -72565965, -74696201, -76771518, -78791971, -80756044, -82663155, -84512076, -86302872, -88034067, -89705316, -91315544, -92865092, -94352657, -95778072, -97140241, -98439585, -99674922, -100846363, -101953029, -102995643, -103973242, -104886114, -105733366, -106515783, -107232507, -107884029, -108469586, -108990146, -109445031, -109834907, -110159075, -110418598, -110612956, -110743021, -110808247, -110809873, -110747573, -110622392, -110433872, -110183341, -109870630, -109496925, -109061853, -108566821, -108011814, -107398172, -106725651, -105995758, -105208649, -104365776, -103466973, -102513780, -101506507, -100446746, -99334505, -98171433, -96958015, -95695913, -94385172, -93027406, -91623219, -90174378, -88681103, -87145094, -85567114, -83948941, -82290788, -80594255, -78860190, -77090470, -75285527, -73447070, -71576122, -69674546, -67742761, -65782346, -63794384, -61780793, -59742181, -57680191, -55596034, -53491542, -51367245, -49224577, -47064766, -44889688, -42700109, -40497558, -38283410, -36059459, -33826411, -31585598, -29338397, -27086596, -24831081, -22573215, -20314456, -18056430, -15799888, -13545910, -11295904, -9051483, -6813612, -4583444, -2362482, -152205, --2046514, --4232760, --6405078, --8562051, --10702661, --12825994, --14930573, --17015184, --19078982, --21121361, --23140947, --25136591, --27107282, --29052375, --30970456, --32860543, --34721731, --36553615, --38354874, --40124623, --41861868, --43566233, --45236425, --46871781, --48471484, --50035443, --51562506, --53052109, --54503334, --55916090, --57289239, --58622387, --59914726, --61166369, --62376298, --63544241, --64669360, --65751822, --66790682, --67785875, --68736719, --69643614, --70505769, --71323237, --72095315, --72822435, --73503872, --74139832, --74729705, --75274073, --75772337, --76224828, --76630970, --76991410, --77305656, --77574197, --77796578, --77973591, --78104888, --78191082, --78231765, --78227787, --78178906, --78085845, --77948266, --77767093, --77542200, --77274423, --76963506, --76610446, --76215234, --75778805, --75300971, --74782775, --74224327, --73626669, --72989720, --72314602, --71601546, --70851655, --70064886, --69242354, --68384382, --67492158, --66565755, --65606354, --64614392, --63591082, --62536507, --61451798, --60337466, --59194800, --58024034, --56826376, --55602460, --54353584, --53079977, --51782773, --50462662, --49120984, --47758105, --46375208, --44973078, --43553013, --42115335, --40661095, --39191102, --37706696, --36208364, --34697230, --33174211, --31640600, --30096851, --28543956, --26982847, --25414822, --23840464, --22260795, --20676812, --19089709, --17499984, --15908468, --14316132, --12724175, --11133249, --9544240, --7958198, --6376234, --4798939, --3227038, --1661555, --103565, -1446230, -2987101, -4517998, -6037984, -7546475, -9042956, -10526443, -11996036, -13451023, -14890857, -16314514, -17721207, -19110300, -20481414, -21833586, -23166092, -24478220, -25769611, -27039309, -28286749, -29511342, -30712939, -31890678, -33044057, -34172405, -35275563, -36352672, -37403349, -38427003, -39423626, -40392440, -41333144, -42245116, -43128383, -43982211, -44806449, -45600591, -46364840, -47098569, -47801711, -48473727, -49114838, -49724455, -50302623, -50848874, -51363545, -51846135, -52296779, -52715020, -53101240, -53455010, -53776585, -54065600, -54322557, -54547131, -54739663, -54899810, -55028111, -55124307, -55188827, -55221380, -55222572, -55192228, -55130859, -55038225, -54914980, -54761033, -54576974, -54362622, -54118682, -53845150, -53542695, -53211202, -52851427, -52463450, -52047991, -51604968, -51135146, -50638673, -50116337, -49568129, -48994861, 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required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "AudioResampler.h" +#include <media/AudioBufferProvider.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <string.h> +#include <sys/mman.h> +#include <sys/stat.h> +#include <errno.h> +#include <time.h> +#include <math.h> + +using namespace android; + +struct HeaderWav { + HeaderWav(size_t size, int nc, int sr, int bits) { + strncpy(RIFF, "RIFF", 4); + chunkSize = size + sizeof(HeaderWav); + strncpy(WAVE, "WAVE", 4); + strncpy(fmt, "fmt ", 4); + fmtSize = 16; + audioFormat = 1; + numChannels = nc; + samplesRate = sr; + byteRate = sr * numChannels * (bits/8); + align = nc*(bits/8); + bitsPerSample = bits; + strncpy(data, "data", 4); + dataSize = size; + } + + char RIFF[4]; // RIFF + uint32_t chunkSize; // File size + char WAVE[4]; // WAVE + char fmt[4]; // fmt\0 + uint32_t fmtSize; // fmt size + uint16_t audioFormat; // 1=PCM + uint16_t numChannels; // num channels + uint32_t samplesRate; // sample rate in hz + uint32_t byteRate; // Bps + uint16_t align; // 2=16-bit mono, 4=16-bit stereo + uint16_t bitsPerSample; // bits per sample + char data[4]; // "data" + uint32_t dataSize; // size +}; + +static int usage(const char* name) { + fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] " + "[-o output-sample-rate] [<input-file>] <output-file>\n", name); + fprintf(stderr," -p enable profiling\n"); + fprintf(stderr," -h create wav file\n"); + fprintf(stderr," -s stereo\n"); + fprintf(stderr," -q resampler quality\n"); + fprintf(stderr," dq : default quality\n"); + fprintf(stderr," lq : low quality\n"); + fprintf(stderr," mq : medium quality\n"); + fprintf(stderr," hq : high quality\n"); + fprintf(stderr," vhq : very high quality\n"); + fprintf(stderr," -i input file sample rate\n"); + fprintf(stderr," -o output file sample rate\n"); + return -1; +} + +int main(int argc, char* argv[]) { + + const char* const progname = argv[0]; + bool profiling = false; + bool writeHeader = false; + int channels = 1; + int input_freq = 0; + int output_freq = 0; + AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; + + int ch; + while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) { + switch (ch) { + case 'p': + profiling = true; + break; + case 'h': + writeHeader = true; + break; + case 's': + channels = 2; + break; + case 'q': + if (!strcmp(optarg, "dq")) + quality = AudioResampler::DEFAULT_QUALITY; + else if (!strcmp(optarg, "lq")) + quality = AudioResampler::LOW_QUALITY; + else if (!strcmp(optarg, "mq")) + quality = AudioResampler::MED_QUALITY; + else if (!strcmp(optarg, "hq")) + quality = AudioResampler::HIGH_QUALITY; + else if (!strcmp(optarg, "vhq")) + quality = AudioResampler::VERY_HIGH_QUALITY; + else { + usage(progname); + return -1; + } + break; + case 'i': + input_freq = atoi(optarg); + break; + case 'o': + output_freq = atoi(optarg); + break; + case '?': + default: + usage(progname); + return -1; + } + } + argc -= optind; + argv += optind; + + const char* file_in = NULL; + const char* file_out = NULL; + if (argc == 1) { + file_out = argv[0]; + } else if (argc == 2) { + file_in = argv[0]; + file_out = argv[1]; + } else { + usage(progname); + return -1; + } + + // ---------------------------------------------------------- + + size_t input_size; + void* input_vaddr; + if (argc == 2) { + struct stat st; + if (stat(file_in, &st) < 0) { + fprintf(stderr, "stat: %s\n", strerror(errno)); + return -1; + } + + int input_fd = open(file_in, O_RDONLY); + if (input_fd < 0) { + fprintf(stderr, "open: %s\n", strerror(errno)); + return -1; + } + + input_size = st.st_size; + input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0); + if (input_vaddr == MAP_FAILED ) { + fprintf(stderr, "mmap: %s\n", strerror(errno)); + return -1; + } + } else { + double k = 1000; // Hz / s + double time = (input_freq / 2) / k; + size_t input_frames = size_t(input_freq * time); + input_size = channels * sizeof(int16_t) * input_frames; + input_vaddr = malloc(input_size); + int16_t* in = (int16_t*)input_vaddr; + for (size_t i=0 ; i<input_frames ; i++) { + double t = double(i) / input_freq; + double y = sin(M_PI * k * t * t); + int16_t yi = floor(y * 32767.0 + 0.5); + for (size_t j=0 ; j<(size_t)channels ; j++) { + in[i*channels + j] = yi / (1+j); + } + } + } + + // ---------------------------------------------------------- + + class Provider: public AudioBufferProvider { + int16_t* mAddr; + size_t mNumFrames; + public: + Provider(const void* addr, size_t size, int channels) { + mAddr = (int16_t*) addr; + mNumFrames = size / (channels*sizeof(int16_t)); + } + virtual status_t getNextBuffer(Buffer* buffer, + int64_t pts = kInvalidPTS) { + buffer->frameCount = mNumFrames; + buffer->i16 = mAddr; + return NO_ERROR; + } + virtual void releaseBuffer(Buffer* buffer) { + } + } provider(input_vaddr, input_size, channels); + + size_t input_frames = input_size / (channels * sizeof(int16_t)); + size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq; + output_size &= ~7; // always stereo, 32-bits + + void* output_vaddr = malloc(output_size); + + if (profiling) { + AudioResampler* resampler = AudioResampler::create(16, channels, + output_freq, quality); + + size_t out_frames = output_size/8; + resampler->setSampleRate(input_freq); + resampler->setVolume(0x1000, 0x1000); + + memset(output_vaddr, 0, output_size); + timespec start, end; + clock_gettime(CLOCK_MONOTONIC_HR, &start); + resampler->resample((int*) output_vaddr, out_frames, &provider); + resampler->resample((int*) output_vaddr, out_frames, &provider); + resampler->resample((int*) output_vaddr, out_frames, &provider); + resampler->resample((int*) output_vaddr, out_frames, &provider); + clock_gettime(CLOCK_MONOTONIC_HR, &end); + int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; + int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; + int64_t time = (end_ns - start_ns)/4; + printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6); + + delete resampler; + } + + AudioResampler* resampler = AudioResampler::create(16, channels, + output_freq, quality); + size_t out_frames = output_size/8; + resampler->setSampleRate(input_freq); + resampler->setVolume(0x1000, 0x1000); + + memset(output_vaddr, 0, output_size); + resampler->resample((int*) output_vaddr, out_frames, &provider); + + // down-mix (we just truncate and keep the left channel) + int32_t* out = (int32_t*) output_vaddr; + int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t)); + for (size_t i = 0; i < out_frames; i++) { + for (int j=0 ; j<channels ; j++) { + int32_t s = out[i * 2 + j] >> 12; + if (s > 32767) s = 32767; + else if (s < -32768) s = -32768; + convert[i * channels + j] = int16_t(s); + } + } + + // write output to disk + int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC, + S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); + if (output_fd < 0) { + fprintf(stderr, "open: %s\n", strerror(errno)); + return -1; + } + + if (writeHeader) { + HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16); + write(output_fd, &wav, sizeof(wav)); + } + + write(output_fd, convert, out_frames * channels * sizeof(int16_t)); + close(output_fd); + + return 0; +} diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk index eff47c8..5245983 100644 --- a/services/camera/libcameraservice/Android.mk +++ b/services/camera/libcameraservice/Android.mk @@ -28,7 +28,6 @@ LOCAL_SHARED_LIBRARIES:= \ libbinder \ libcutils \ libmedia \ - libmedia_native \ libcamera_client \ libgui \ libhardware \ @@ -40,6 +39,9 @@ LOCAL_C_INCLUDES += \ system/media/camera/include \ external/jpeg + +LOCAL_CFLAGS += -Wall -Wextra + LOCAL_MODULE:= libcameraservice include $(BUILD_SHARED_LIBRARY) diff --git a/services/camera/libcameraservice/Camera2Client.cpp b/services/camera/libcameraservice/Camera2Client.cpp index e59a240..9627416 100644 --- a/services/camera/libcameraservice/Camera2Client.cpp +++ b/services/camera/libcameraservice/Camera2Client.cpp @@ -37,10 +37,6 @@ static int getCallingPid() { return IPCThreadState::self()->getCallingPid(); } -static int getCallingUid() { - return IPCThreadState::self()->getCallingUid(); -} - // Interface used by CameraService Camera2Client::Camera2Client(const sp<CameraService>& cameraService, @@ -370,7 +366,6 @@ status_t Camera2Client::dump(int fd, const Vector<String16>& args) { void Camera2Client::disconnect() { ATRACE_CALL(); Mutex::Autolock icl(mICameraLock); - status_t res; // Allow both client and the media server to disconnect at all times int callingPid = getCallingPid(); @@ -575,7 +570,7 @@ void Camera2Client::setPreviewCallbackFlag(int flag) { ATRACE_CALL(); ALOGV("%s: Camera %d: Flag 0x%x", __FUNCTION__, mCameraId, flag); Mutex::Autolock icl(mICameraLock); - status_t res; + if ( checkPid(__FUNCTION__) != OK) return; SharedParameters::Lock l(mParameters); @@ -1062,7 +1057,7 @@ status_t Camera2Client::cancelAutoFocus() { return OK; } -status_t Camera2Client::takePicture(int msgType) { +status_t Camera2Client::takePicture(int /*msgType*/) { ATRACE_CALL(); Mutex::Autolock icl(mICameraLock); status_t res; @@ -1244,7 +1239,7 @@ status_t Camera2Client::commandPlayRecordingSoundL() { return OK; } -status_t Camera2Client::commandStartFaceDetectionL(int type) { +status_t Camera2Client::commandStartFaceDetectionL(int /*type*/) { ALOGV("%s: Camera %d: Starting face detection", __FUNCTION__, mCameraId); status_t res; @@ -1331,6 +1326,8 @@ void Camera2Client::notifyError(int errorCode, int arg1, int arg2) { } void Camera2Client::notifyShutter(int frameNumber, nsecs_t timestamp) { + (void)frameNumber; + (void)timestamp; ALOGV("%s: Shutter notification for frame %d at time %lld", __FUNCTION__, frameNumber, timestamp); } @@ -1452,6 +1449,8 @@ void Camera2Client::notifyAutoExposure(uint8_t newState, int triggerId) { } void Camera2Client::notifyAutoWhitebalance(uint8_t newState, int triggerId) { + (void)newState; + (void)triggerId; ALOGV("%s: Auto-whitebalance state now %d, last trigger %d", __FUNCTION__, newState, triggerId); } diff --git a/services/camera/libcameraservice/Camera2Device.cpp b/services/camera/libcameraservice/Camera2Device.cpp index d6445c1..5bfa085 100644 --- a/services/camera/libcameraservice/Camera2Device.cpp +++ b/services/camera/libcameraservice/Camera2Device.cpp @@ -765,7 +765,6 @@ status_t Camera2Device::MetadataQueue::setStreamSlot( ATRACE_CALL(); ALOGV("%s: E", __FUNCTION__); Mutex::Autolock l(mMutex); - status_t res; if (mStreamSlotCount > 0) { freeBuffers(mStreamSlot.begin(), mStreamSlot.end()); @@ -785,7 +784,7 @@ status_t Camera2Device::MetadataQueue::setStreamSlot( } status_t Camera2Device::MetadataQueue::dump(int fd, - const Vector<String16>& args) { + const Vector<String16>& /*args*/) { ATRACE_CALL(); String8 result; status_t notLocked; @@ -894,12 +893,13 @@ int Camera2Device::MetadataQueue::consumer_free( { ATRACE_CALL(); MetadataQueue *queue = getInstance(q); + (void)queue; free_camera_metadata(old_buffer); return OK; } int Camera2Device::MetadataQueue::producer_dequeue( - const camera2_frame_queue_dst_ops_t *q, + const camera2_frame_queue_dst_ops_t * /*q*/, size_t entries, size_t bytes, camera_metadata_t **buffer) { @@ -912,7 +912,7 @@ int Camera2Device::MetadataQueue::producer_dequeue( } int Camera2Device::MetadataQueue::producer_cancel( - const camera2_frame_queue_dst_ops_t *q, + const camera2_frame_queue_dst_ops_t * /*q*/, camera_metadata_t *old_buffer) { ATRACE_CALL(); @@ -1184,7 +1184,7 @@ status_t Camera2Device::StreamAdapter::setTransform(int transform) { } status_t Camera2Device::StreamAdapter::dump(int fd, - const Vector<String16>& args) { + const Vector<String16>& /*args*/) { ATRACE_CALL(); String8 result = String8::format(" Stream %d: %d x %d, format 0x%x\n", mId, mWidth, mHeight, mFormat); @@ -1423,7 +1423,7 @@ status_t Camera2Device::ReprocessStreamAdapter::pushIntoStream( } status_t Camera2Device::ReprocessStreamAdapter::dump(int fd, - const Vector<String16>& args) { + const Vector<String16>& /*args*/) { ATRACE_CALL(); String8 result = String8::format(" Reprocess stream %d: %d x %d, fmt 0x%x\n", @@ -1444,7 +1444,7 @@ int Camera2Device::ReprocessStreamAdapter::acquire_buffer( const camera2_stream_in_ops_t *w, buffer_handle_t** buffer) { ATRACE_CALL(); - int res; + ReprocessStreamAdapter* stream = const_cast<ReprocessStreamAdapter*>( static_cast<const ReprocessStreamAdapter*>(w)); diff --git a/services/camera/libcameraservice/CameraClient.cpp b/services/camera/libcameraservice/CameraClient.cpp index b930c02..006a9c9 100644 --- a/services/camera/libcameraservice/CameraClient.cpp +++ b/services/camera/libcameraservice/CameraClient.cpp @@ -34,10 +34,6 @@ static int getCallingPid() { return IPCThreadState::self()->getCallingPid(); } -static int getCallingUid() { - return IPCThreadState::self()->getCallingUid(); -} - CameraClient::CameraClient(const sp<CameraService>& cameraService, const sp<ICameraClient>& cameraClient, int cameraId, int cameraFacing, int clientPid, int servicePid): diff --git a/services/camera/libcameraservice/CameraHardwareInterface.h b/services/camera/libcameraservice/CameraHardwareInterface.h index 05ac9fa..167b37c 100644 --- a/services/camera/libcameraservice/CameraHardwareInterface.h +++ b/services/camera/libcameraservice/CameraHardwareInterface.h @@ -427,7 +427,7 @@ public: /** * Dump state of the camera hardware */ - status_t dump(int fd, const Vector<String16>& args) const + status_t dump(int fd, const Vector<String16>& /*args*/) const { ALOGV("%s(%s)", __FUNCTION__, mName.string()); if (mDevice->ops->dump) @@ -584,9 +584,10 @@ private: #endif static int __lock_buffer(struct preview_stream_ops* w, - buffer_handle_t* buffer) + buffer_handle_t* /*buffer*/) { ANativeWindow *a = anw(w); + (void)a; return 0; } diff --git a/services/camera/libcameraservice/camera2/BurstCapture.cpp b/services/camera/libcameraservice/camera2/BurstCapture.cpp index f56c50c..192d419 100644 --- a/services/camera/libcameraservice/camera2/BurstCapture.cpp +++ b/services/camera/libcameraservice/camera2/BurstCapture.cpp @@ -38,7 +38,8 @@ BurstCapture::BurstCapture(wp<Camera2Client> client, wp<CaptureSequencer> sequen BurstCapture::~BurstCapture() { } -status_t BurstCapture::start(Vector<CameraMetadata> &metadatas, int32_t firstCaptureId) { +status_t BurstCapture::start(Vector<CameraMetadata> &/*metadatas*/, + int32_t /*firstCaptureId*/) { ALOGE("Not completely implemented"); return INVALID_OPERATION; } @@ -75,7 +76,7 @@ bool BurstCapture::threadLoop() { CpuConsumer::LockedBuffer* BurstCapture::jpegEncode( CpuConsumer::LockedBuffer *imgBuffer, - int quality) + int /*quality*/) { ALOGV("%s", __FUNCTION__); @@ -91,7 +92,7 @@ CpuConsumer::LockedBuffer* BurstCapture::jpegEncode( buffers.push_back(imgEncoded); sp<JpegCompressor> jpeg = new JpegCompressor(); - status_t res = jpeg->start(buffers, 1); + jpeg->start(buffers, 1); bool success = jpeg->waitForDone(10 * 1e9); if(success) { @@ -103,7 +104,7 @@ CpuConsumer::LockedBuffer* BurstCapture::jpegEncode( } } -status_t BurstCapture::processFrameAvailable(sp<Camera2Client> &client) { +status_t BurstCapture::processFrameAvailable(sp<Camera2Client> &/*client*/) { ALOGE("Not implemented"); return INVALID_OPERATION; } diff --git a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp index 3e9c255..307cfab 100644 --- a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp +++ b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp @@ -119,7 +119,6 @@ status_t CallbackProcessor::updateStream(const Parameters ¶ms) { status_t CallbackProcessor::deleteStream() { ATRACE_CALL(); - status_t res; Mutex::Autolock l(mInputMutex); @@ -144,7 +143,7 @@ int CallbackProcessor::getStreamId() const { return mCallbackStreamId; } -void CallbackProcessor::dump(int fd, const Vector<String16>& args) const { +void CallbackProcessor::dump(int /*fd*/, const Vector<String16>& /*args*/) const { } bool CallbackProcessor::threadLoop() { @@ -173,7 +172,6 @@ status_t CallbackProcessor::processNewCallback(sp<Camera2Client> &client) { ATRACE_CALL(); status_t res; - int callbackHeapId; sp<Camera2Heap> callbackHeap; size_t heapIdx; diff --git a/services/camera/libcameraservice/camera2/CaptureSequencer.cpp b/services/camera/libcameraservice/camera2/CaptureSequencer.cpp index fe4abc0..b228faf 100644 --- a/services/camera/libcameraservice/camera2/CaptureSequencer.cpp +++ b/services/camera/libcameraservice/camera2/CaptureSequencer.cpp @@ -128,7 +128,7 @@ void CaptureSequencer::onCaptureAvailable(nsecs_t timestamp, } -void CaptureSequencer::dump(int fd, const Vector<String16>& args) { +void CaptureSequencer::dump(int fd, const Vector<String16>& /*args*/) { String8 result; if (mCaptureRequest.entryCount() != 0) { result = " Capture request:\n"; @@ -182,7 +182,6 @@ const CaptureSequencer::StateManager }; bool CaptureSequencer::threadLoop() { - status_t res; sp<Camera2Client> client = mClient.promote(); if (client == 0) return false; @@ -213,7 +212,8 @@ bool CaptureSequencer::threadLoop() { return true; } -CaptureSequencer::CaptureState CaptureSequencer::manageIdle(sp<Camera2Client> &client) { +CaptureSequencer::CaptureState CaptureSequencer::manageIdle( + sp<Camera2Client> &/*client*/) { status_t res; Mutex::Autolock l(mInputMutex); while (!mStartCapture) { @@ -350,13 +350,13 @@ CaptureSequencer::CaptureState CaptureSequencer::manageZslStart( } CaptureSequencer::CaptureState CaptureSequencer::manageZslWaiting( - sp<Camera2Client> &client) { + sp<Camera2Client> &/*client*/) { ALOGV("%s", __FUNCTION__); return DONE; } CaptureSequencer::CaptureState CaptureSequencer::manageZslReprocessing( - sp<Camera2Client> &client) { + sp<Camera2Client> &/*client*/) { ALOGV("%s", __FUNCTION__); return START; } @@ -378,7 +378,7 @@ CaptureSequencer::CaptureState CaptureSequencer::manageStandardStart( } CaptureSequencer::CaptureState CaptureSequencer::manageStandardPrecaptureWait( - sp<Camera2Client> &client) { + sp<Camera2Client> &/*client*/) { status_t res; ATRACE_CALL(); Mutex::Autolock l(mInputMutex); @@ -578,7 +578,7 @@ CaptureSequencer::CaptureState CaptureSequencer::manageBurstCaptureStart( } CaptureSequencer::CaptureState CaptureSequencer::manageBurstCaptureWait( - sp<Camera2Client> &client) { + sp<Camera2Client> &/*client*/) { status_t res; ATRACE_CALL(); diff --git a/services/camera/libcameraservice/camera2/FrameProcessor.cpp b/services/camera/libcameraservice/camera2/FrameProcessor.cpp index 064607c..e032522 100644 --- a/services/camera/libcameraservice/camera2/FrameProcessor.cpp +++ b/services/camera/libcameraservice/camera2/FrameProcessor.cpp @@ -62,7 +62,7 @@ status_t FrameProcessor::removeListener(int32_t minId, return OK; } -void FrameProcessor::dump(int fd, const Vector<String16>& args) { +void FrameProcessor::dump(int fd, const Vector<String16>& /*args*/) { String8 result(" Latest received frame:\n"); write(fd, result.string(), result.size()); mLastFrame.dump(fd, 2, 6); @@ -128,7 +128,6 @@ void FrameProcessor::processNewFrames(sp<Camera2Client> &client) { status_t FrameProcessor::processListeners(const CameraMetadata &frame, sp<Camera2Client> &client) { - status_t res; ATRACE_CALL(); camera_metadata_ro_entry_t entry; @@ -173,7 +172,7 @@ status_t FrameProcessor::processFaceDetect(const CameraMetadata &frame, ATRACE_CALL(); camera_metadata_ro_entry_t entry; bool enableFaceDetect; - int maxFaces; + { SharedParameters::Lock l(client->getParameters()); enableFaceDetect = l.mParameters.enableFaceDetect; diff --git a/services/camera/libcameraservice/camera2/JpegCompressor.cpp b/services/camera/libcameraservice/camera2/JpegCompressor.cpp index 702ef58..c9af71e 100644 --- a/services/camera/libcameraservice/camera2/JpegCompressor.cpp +++ b/services/camera/libcameraservice/camera2/JpegCompressor.cpp @@ -144,7 +144,7 @@ bool JpegCompressor::isBusy() { } // old function -- TODO: update for new buffer type -bool JpegCompressor::isStreamInUse(uint32_t id) { +bool JpegCompressor::isStreamInUse(uint32_t /*id*/) { ALOGV("%s", __FUNCTION__); Mutex::Autolock lock(mBusyMutex); @@ -203,14 +203,14 @@ void JpegCompressor::jpegInitDestination(j_compress_ptr cinfo) { dest->free_in_buffer = kMaxJpegSize; } -boolean JpegCompressor::jpegEmptyOutputBuffer(j_compress_ptr cinfo) { +boolean JpegCompressor::jpegEmptyOutputBuffer(j_compress_ptr /*cinfo*/) { ALOGV("%s", __FUNCTION__); ALOGE("%s: JPEG destination buffer overflow!", __FUNCTION__); return true; } -void JpegCompressor::jpegTermDestination(j_compress_ptr cinfo) { +void JpegCompressor::jpegTermDestination(j_compress_ptr /*cinfo*/) { ALOGV("%s", __FUNCTION__); ALOGV("%s: Done writing JPEG data. %d bytes left in buffer", __FUNCTION__, cinfo->dest->free_in_buffer); diff --git a/services/camera/libcameraservice/camera2/JpegProcessor.cpp b/services/camera/libcameraservice/camera2/JpegProcessor.cpp index ffc072b..6280f83 100644 --- a/services/camera/libcameraservice/camera2/JpegProcessor.cpp +++ b/services/camera/libcameraservice/camera2/JpegProcessor.cpp @@ -139,7 +139,6 @@ status_t JpegProcessor::updateStream(const Parameters ¶ms) { status_t JpegProcessor::deleteStream() { ATRACE_CALL(); - status_t res; Mutex::Autolock l(mInputMutex); @@ -164,7 +163,7 @@ int JpegProcessor::getStreamId() const { return mCaptureStreamId; } -void JpegProcessor::dump(int fd, const Vector<String16>& args) const { +void JpegProcessor::dump(int /*fd*/, const Vector<String16>& /*args*/) const { } bool JpegProcessor::threadLoop() { @@ -356,7 +355,7 @@ size_t JpegProcessor::findJpegSize(uint8_t* jpegBuffer, size_t maxSize) { // Find End of Image // Scan JPEG buffer until End of Image (EOI) bool foundEnd = false; - for (size; size <= maxSize - MARKER_LENGTH; size++) { + for ( ; size <= maxSize - MARKER_LENGTH; size++) { if ( checkJpegEnd(jpegBuffer + size) ) { foundEnd = true; size += MARKER_LENGTH; diff --git a/services/camera/libcameraservice/camera2/Parameters.cpp b/services/camera/libcameraservice/camera2/Parameters.cpp index 9a0083a..93927e6 100644 --- a/services/camera/libcameraservice/camera2/Parameters.cpp +++ b/services/camera/libcameraservice/camera2/Parameters.cpp @@ -951,7 +951,6 @@ status_t Parameters::buildQuirks() { camera_metadata_ro_entry_t Parameters::staticInfo(uint32_t tag, size_t minCount, size_t maxCount) const { - status_t res; camera_metadata_ro_entry_t entry = info->find(tag); if (CC_UNLIKELY( entry.count == 0 )) { @@ -1567,6 +1566,10 @@ status_t Parameters::set(const String8& paramString) { ALOGE("%s: Video stabilization not supported", __FUNCTION__); } + // LIGHTFX + validatedParams.lightFx = lightFxStringToEnum( + newParams.get(CameraParameters::KEY_LIGHTFX)); + /** Update internal parameters */ *this = validatedParams; @@ -2094,6 +2097,18 @@ const char *Parameters::focusModeEnumToString(focusMode_t focusMode) { } } +Parameters::Parameters::lightFxMode_t Parameters::lightFxStringToEnum( + const char *lightFxMode) { + return + !lightFxMode ? + Parameters::LIGHTFX_NONE : + !strcmp(lightFxMode, CameraParameters::LIGHTFX_LOWLIGHT) ? + Parameters::LIGHTFX_LOWLIGHT : + !strcmp(lightFxMode, CameraParameters::LIGHTFX_HDR) ? + Parameters::LIGHTFX_HDR : + Parameters::LIGHTFX_NONE; +} + status_t Parameters::parseAreas(const char *areasCStr, Vector<Parameters::Area> *areas) { static const size_t NUM_FIELDS = 5; @@ -2414,7 +2429,7 @@ Parameters::CropRegion Parameters::calculateCropRegion( return crop; } -int32_t Parameters::fpsFromRange(int32_t min, int32_t max) const { +int32_t Parameters::fpsFromRange(int32_t /*min*/, int32_t max) const { return max; } diff --git a/services/camera/libcameraservice/camera2/Parameters.h b/services/camera/libcameraservice/camera2/Parameters.h index 54b1e8c..6d32bf6 100644 --- a/services/camera/libcameraservice/camera2/Parameters.h +++ b/services/camera/libcameraservice/camera2/Parameters.h @@ -261,6 +261,8 @@ struct Parameters { static const char* flashModeEnumToString(flashMode_t flashMode); static focusMode_t focusModeStringToEnum(const char *focusMode); static const char* focusModeEnumToString(focusMode_t focusMode); + static lightFxMode_t lightFxStringToEnum(const char *lightFxMode); + static status_t parseAreas(const char *areasCStr, Vector<Area> *areas); diff --git a/services/camera/libcameraservice/camera2/StreamingProcessor.cpp b/services/camera/libcameraservice/camera2/StreamingProcessor.cpp index 207f780..6ea27b2 100644 --- a/services/camera/libcameraservice/camera2/StreamingProcessor.cpp +++ b/services/camera/libcameraservice/camera2/StreamingProcessor.cpp @@ -447,7 +447,6 @@ status_t StreamingProcessor::incrementStreamingIds() { ATRACE_CALL(); Mutex::Autolock m(mMutex); - status_t res; mPreviewRequestId++; if (mPreviewRequestId >= Camera2Client::kPreviewRequestIdEnd) { mPreviewRequestId = Camera2Client::kPreviewRequestIdStart; @@ -628,7 +627,7 @@ void StreamingProcessor::releaseRecordingFrame(const sp<IMemory>& mem) { } -status_t StreamingProcessor::dump(int fd, const Vector<String16>& args) { +status_t StreamingProcessor::dump(int fd, const Vector<String16>& /*args*/) { String8 result; result.append(" Current requests:\n"); diff --git a/services/camera/libcameraservice/camera2/ZslProcessor.cpp b/services/camera/libcameraservice/camera2/ZslProcessor.cpp index 1937955..9584028 100644 --- a/services/camera/libcameraservice/camera2/ZslProcessor.cpp +++ b/services/camera/libcameraservice/camera2/ZslProcessor.cpp @@ -69,11 +69,12 @@ void ZslProcessor::onFrameAvailable() { } } -void ZslProcessor::onFrameAvailable(int32_t frameId, const CameraMetadata &frame) { +void ZslProcessor::onFrameAvailable(int32_t /*frameId*/, const CameraMetadata &frame) { Mutex::Autolock l(mInputMutex); camera_metadata_ro_entry_t entry; entry = frame.find(ANDROID_SENSOR_TIMESTAMP); nsecs_t timestamp = entry.data.i64[0]; + (void)timestamp; ALOGVV("Got preview frame for timestamp %lld", timestamp); if (mState != RUNNING) return; @@ -367,7 +368,7 @@ status_t ZslProcessor::clearZslQueueLocked() { return OK; } -void ZslProcessor::dump(int fd, const Vector<String16>& args) const { +void ZslProcessor::dump(int fd, const Vector<String16>& /*args*/) const { Mutex::Autolock l(mInputMutex); if (!mLatestCapturedRequest.isEmpty()) { String8 result(" Latest ZSL capture request:\n"); diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp index 377814f..cc3d509 100644 --- a/tools/resampler_tools/fir.cpp +++ b/tools/resampler_tools/fir.cpp @@ -16,6 +16,9 @@ #include <math.h> #include <stdio.h> +#include <unistd.h> +#include <stdlib.h> +#include <string.h> static double sinc(double x) { if (fabs(x) == 0.0f) return 1.0f; @@ -34,44 +37,82 @@ static double I0(double x) { y=x/3.75; y*=y; ans=1.0+y*(3.5156229+y*(3.0899424+y*(1.2067492 - +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2))))); + +y*(0.2659732+y*(0.360768e-1+y*0.45813e-2))))); } else { y=3.75/ax; ans=(exp(ax)/sqrt(ax))*(0.39894228+y*(0.1328592e-1 - +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2 - +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1 - +y*0.392377e-2)))))))); + +y*(0.225319e-2+y*(-0.157565e-2+y*(0.916281e-2 + +y*(-0.2057706e-1+y*(0.2635537e-1+y*(-0.1647633e-1 + +y*0.392377e-2)))))))); } return ans; } -static double kaiser(int k, int N, double alpha) { +static double kaiser(int k, int N, double beta) { if (k < 0 || k > N) return 0; - return I0(M_PI*alpha * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(M_PI*alpha); + return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta); +} + + +static void usage(char* name) { + fprintf(stderr, + "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] [-l lerp]\n" + " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] -p M/N\n" + " -h this help message\n" + " -d debug, print comma-separated coefficient table\n" + " -p generate poly-phase filter coefficients, with sample increment M/N\n" + " -s sample rate (48000)\n" + " -c cut-off frequency (20478)\n" + " -n number of zero-crossings on one side (8)\n" + " -l number of lerping bits (4)\n" + " -f output format, can be fixed-point or floating-point (fixed)\n" + " -b kaiser window parameter beta (7.865 [-80dB])\n" + " -v attenuation in dBFS (0)\n", + name, name + ); + exit(0); } int main(int argc, char** argv) { // nc is the number of bits to store the coefficients - int nc = 32; + const int nc = 32; - // ni is the minimum number of bits needed for interpolation - // (not used for generating the coefficients) - const int ni = nc / 2; + bool polyphase = false; + unsigned int polyM = 160; + unsigned int polyN = 147; + bool debug = false; + double Fs = 48000; + double Fc = 20478; + double atten = 1; + int format = 0; - // cut off frequency ratio Fc/Fs - // The bigger the stop-band, the less coefficients we'll need. - double Fcr = 20000.0 / 48000.0; - // nzc is the number of zero-crossing on one half of the filter - int nzc = 8; - - // alpha parameter of the kaiser window - // Larger numbers reduce ripples in the rejection band but increase - // the width of the transition band. - // the table below gives some value of alpha for a given - // stop-band attenuation. + // in order to keep the errors associated with the linear + // interpolation of the coefficients below the quantization error + // we must satisfy: + // 2^nz >= 2^(nc/2) + // + // for 16 bit coefficients that would be 256 + // + // note that increasing nz only increases memory requirements, + // but doesn't increase the amount of computation to do. + // + // + // see: + // Smith, J.O. Digital Audio Resampling Home Page + // https://ccrma.stanford.edu/~jos/resample/, 2011-03-29 + // + int nz = 4; + + // | 0.1102*(A - 8.7) A > 50 + // beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50 + // | 0 A < 21 + // with A is the desired stop-band attenuation in dBFS + // + // for eg: + // // 30 dB 2.210 // 40 dB 3.384 // 50 dB 4.538 @@ -80,42 +121,161 @@ int main(int argc, char** argv) // 80 dB 7.865 // 90 dB 8.960 // 100 dB 10.056 - double alpha = 7.865; // -80dB stop-band attenuation - - // 2^nz is the number coefficients per zero-crossing - // (int theory this should be 1<<(nc/2)) - const int nz = 4; + double beta = 7.865; + + + // 2*nzc = (A - 8) / (2.285 * dw) + // with dw the transition width = 2*pi*dF/Fs + // + int nzc = 8; + + // + // Example: + // 44.1 KHz to 48 KHz resampling + // 100 dB rejection above 28 KHz + // (the spectrum will fold around 24 KHz and we want 100 dB rejection + // at the point where the folding reaches 20 KHz) + // ...___|_____ + // | \| + // | ____/|\____ + // |/alias| \ + // ------/------+------\---------> KHz + // 20 24 28 + + // Transition band 8 KHz, or dw = 1.0472 + // + // beta = 10.056 + // nzc = 20 + // + + int ch; + while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) { + switch (ch) { + case 'd': + debug = true; + break; + case 'p': + if (sscanf(optarg, "%u/%u", &polyM, &polyN) != 2) { + usage(argv[0]); + } + polyphase = true; + break; + case 's': + Fs = atof(optarg); + break; + case 'c': + Fc = atof(optarg); + break; + case 'n': + nzc = atoi(optarg); + break; + case 'l': + nz = atoi(optarg); + break; + case 'f': + if (!strcmp(optarg,"fixed")) format = 0; + else if (!strcmp(optarg,"float")) format = 1; + else usage(argv[0]); + break; + case 'b': + beta = atof(optarg); + break; + case 'v': + atten = pow(10, -fabs(atof(optarg))*0.05 ); + break; + case 'h': + default: + usage(argv[0]); + break; + } + } + + // cut off frequency ratio Fc/Fs + double Fcr = Fc / Fs; + - // total number of coefficients - const int N = (1 << nz) * nzc; + // total number of coefficients (one side) + const int M = (1 << nz); + const int N = M * nzc; // generate the right half of the filter - printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N); - printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc); - printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc); - printf("const int32_t RESAMPLE_FIR_LERP_FRAC_BITS = %d;\n", ni); - printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS = %d;\n", nz); - printf("\n"); - printf("static int16_t resampleFIR[%d] = {", N); - for (int i=0 ; i<N ; i++) - { - double x = (2.0 * M_PI * i * Fcr) / (1 << nz); - double y = kaiser(i+N, 2*N, alpha) * sinc(x); - - long yi = floor(y * ((1ULL<<(nc-1))) + 0.5); - if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1; - - if ((i % (1 << 4)) == 0) printf("\n "); - if (nc > 16) - printf("0x%08x, ", int(yi)); - else - printf("0x%04x, ", int(yi)&0xFFFF); + if (!debug) { + printf("// cmd-line: "); + for (int i=1 ; i<argc ; i++) { + printf("%s ", argv[i]); + } + printf("\n"); + if (!polyphase) { + printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N); + printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS = %d;\n", nz); + printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc); + } else { + printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN); + printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", 2*nzc); + } + if (!format) { + printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc); + } + printf("\n"); + printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float"); + } + + if (!polyphase) { + for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation + for (int j=0 ; j<nzc ; j++) { + int ix = j*M + i; + double x = (2.0 * M_PI * ix * Fcr) / (1 << nz); + double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr; + y *= atten; + + if (!debug) { + if (j == 0) + printf("\n "); + } + + if (!format) { + int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5); + if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1; + printf("0x%08x, ", int32_t(yi)); + } else { + printf("%.9g%s ", y, debug ? "," : "f,"); + } + } + } + } else { + for (int j=0 ; j<polyN ; j++) { + // calculate the phase + double p = ((polyM*j) % polyN) / double(polyN); + if (!debug) printf("\n "); + else printf("\n"); + // generate a FIR per phase + for (int i=-nzc ; i<nzc ; i++) { + double x = 2.0 * M_PI * Fcr * (i + p); + double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;; + y *= atten; + if (!format) { + int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5); + if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1; + printf("0x%08x", int32_t(yi)); + } else { + printf("%.9g%s", y, debug ? "" : "f"); + } + + if (debug && (i==nzc-1)) { + } else { + printf(", "); + } + } + } + } + + if (!debug) { + printf("\n};"); } - printf("\n};\n"); + printf("\n"); return 0; - } +} -// http://www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html // http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html - + |