diff options
70 files changed, 1252 insertions, 620 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/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/AudioSystem.h b/include/media/AudioSystem.h index 49e1afc..2218fad 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -87,9 +87,12 @@ 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(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 getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, int* samplingRate); @@ -126,7 +129,8 @@ 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(uint32_t *halFrames, uint32_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); @@ -147,8 +151,8 @@ 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 +166,8 @@ 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 +187,10 @@ 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); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 34108b3..7dd22e8 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -53,9 +53,12 @@ public: 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_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. }; @@ -312,7 +315,8 @@ 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 +334,8 @@ 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 +364,8 @@ 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); @@ -518,8 +524,10 @@ protected: 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 + 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; 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..359780e 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -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; 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/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 5b133f3..fe42afa 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -27,7 +27,8 @@ 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 @@ -100,7 +101,8 @@ public: 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: 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/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 --- 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 --- 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 --- 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..bdbee0d 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -127,7 +127,8 @@ status_t AudioRecord::set( int sessionId) { - ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount); + ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, + frameCount); AutoMutex lock(mLock); @@ -701,7 +702,8 @@ 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; diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 207f96f..767c452 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -246,7 +246,8 @@ 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 %d", streamType, output, + *samplingRate); return NO_ERROR; } @@ -290,7 +291,8 @@ 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 +371,8 @@ 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(uint32_t *halFrames, uint32_t *dspFrames, + audio_stream_type_t stream) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; @@ -449,8 +452,10 @@ 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 %d, format %d channels %#x frameCount %d " + "latency %d", + outputDesc->samplingRate, outputDesc->format, outputDesc->channels, + outputDesc->frameCount, outputDesc->latency); } break; case OUTPUT_CLOSED: { if (gOutputs.indexOfKey(ioHandle) < 0) { @@ -471,7 +476,8 @@ 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 %d, 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 +516,7 @@ 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(); diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 362d022..5fc9b07 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -198,7 +198,8 @@ status_t AudioTrack::set( int sessionId) { - 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); @@ -617,12 +618,14 @@ 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; } @@ -924,7 +927,8 @@ status_t AudioTrack::createTrack_l( mCblk->stepUser(mCblk->frameCount); } - mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); + mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | + uint16_t(mVolume[LEFT] * 0x1000)); mCblk->setSendLevel(mSendLevel); mAudioTrack->attachAuxEffect(mAuxEffectId); mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; @@ -994,8 +998,8 @@ 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(); @@ -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; @@ -1439,11 +1444,14 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const String8 result; 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, mCblk->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(%d), status(%d), muted(%d)\n", + (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); result.append(buffer); snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); result.append(buffer); diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index ce8ffc4..f412591 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -865,7 +865,8 @@ 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; @@ -1043,7 +1044,8 @@ 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/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/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/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/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/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/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp index 6b1abb1..ddd5b84 100644 --- a/media/mediaserver/main_mediaserver.cpp +++ b/media/mediaserver/main_mediaserver.cpp @@ -33,6 +33,7 @@ using namespace android; int main(int argc, char** argv) { + signal(SIGPIPE, SIG_IGN); sp<ProcessState> proc(ProcessState::self()); sp<IServiceManager> sm = defaultServiceManager(); ALOGI("ServiceManager: %p", sm.get()); 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..35bd431 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; @@ -941,8 +947,8 @@ 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 +1118,8 @@ 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 +1222,8 @@ 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 +1252,8 @@ 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; @@ -1661,7 +1670,8 @@ 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); @@ -1791,7 +1801,7 @@ 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 %d format %d, channelMask 0x%08x " "for output %p with format %d", sampleRate, format, channelMask, mOutput, mFormat); lStatus = BAD_VALUE; @@ -1959,7 +1969,8 @@ 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 +2036,8 @@ 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 +2045,8 @@ 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 +2075,8 @@ 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 +2095,10 @@ 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 +2112,8 @@ 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]; @@ -2235,7 +2251,8 @@ 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)) { @@ -2891,7 +2908,8 @@ 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 } @@ -3125,7 +3143,8 @@ 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 +3157,8 @@ 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 +3288,8 @@ 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. @@ -3362,7 +3383,8 @@ 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 +3397,8 @@ 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)); } @@ -3580,39 +3603,18 @@ 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]; @@ -3660,6 +3662,34 @@ 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 +3761,7 @@ 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 +3822,7 @@ 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 +4012,8 @@ 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); @@ -4103,18 +4134,21 @@ 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; } } @@ -4161,7 +4195,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( // mChannelCount // mChannelMask { - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + 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); @@ -4322,7 +4357,8 @@ AudioFlinger::PlaybackThread::Track::Track( 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 @@ -4341,7 +4377,8 @@ AudioFlinger::PlaybackThread::Track::Track( 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); + 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; @@ -4366,7 +4403,8 @@ 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 +4446,8 @@ 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 mFrCnt 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) @@ -4636,7 +4674,8 @@ 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(); @@ -5395,7 +5434,8 @@ 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; @@ -5587,7 +5627,8 @@ 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; } @@ -5599,7 +5640,8 @@ 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); pInBuffer->frameCount -= outFrames; @@ -5612,7 +5654,8 @@ 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 +5671,14 @@ 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); } } } @@ -5657,14 +5703,15 @@ 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(); @@ -5921,13 +5968,14 @@ 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 +5994,8 @@ Exit: // ---------------------------------------------------------------------------- -AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { @@ -5961,7 +6010,8 @@ 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 +6038,21 @@ 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 +6159,8 @@ 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->mCblk->frameSize; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; @@ -6125,14 +6179,17 @@ 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 +6202,9 @@ 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 +6216,14 @@ 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 { @@ -6635,7 +6697,8 @@ 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 +6720,8 @@ 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 +6731,10 @@ 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 +6828,8 @@ 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; } @@ -6989,7 +7056,8 @@ 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 +7110,8 @@ 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 +7146,8 @@ 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 +7234,17 @@ 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 +7255,66 @@ 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 +8122,8 @@ 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 +8714,7 @@ 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 +8965,15 @@ 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 +9057,8 @@ 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 +9173,8 @@ 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 +9185,9 @@ 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 +9327,8 @@ 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 +9487,8 @@ 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 +9539,8 @@ 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 +9571,8 @@ 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 +9796,8 @@ 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 +9823,8 @@ 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..2251b45 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -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(); @@ -746,7 +748,8 @@ 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 { @@ -788,7 +791,8 @@ 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 +814,7 @@ private: // implement FastMixerState::VolumeProvider interface virtual uint32_t getVolumeLR(); + virtual status_t setSyncEvent(const sp<SyncEvent>& event); protected: @@ -822,7 +827,8 @@ 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; @@ -876,8 +882,8 @@ 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 private: IAudioFlinger::track_flags_t mFlags; @@ -986,7 +992,7 @@ private: class OutputTrack : public Track { public: - class Buffer: public AudioBufferProvider::Buffer { + class Buffer : public AudioBufferProvider::Buffer { public: int16_t *mBuffer; }; @@ -999,7 +1005,8 @@ private: int 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 +1020,8 @@ 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() @@ -1025,8 +1033,8 @@ private: DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() }; // 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); @@ -1185,7 +1193,8 @@ 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 +1257,11 @@ 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 +1314,8 @@ 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 +1335,11 @@ 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 +1348,8 @@ 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 @@ -1453,7 +1462,8 @@ 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); @@ -1465,7 +1475,8 @@ 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 +1487,8 @@ 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 ? @@ -1572,6 +1584,9 @@ 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 +1622,7 @@ 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 +1796,8 @@ 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 +1810,11 @@ 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 +2080,13 @@ 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..e0ea4a4 100644 --- a/services/audioflinger/AudioResamplerSinc.cpp +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -31,36 +31,33 @@ 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, + 0x6d374bc7, 0x6d35278a, 0x6d2ebafe, 0x6d24069d, 0x6d150b35, 0x6d01c9e3, 0x6cea4418, 0x6cce7b97, 0x6cae7272, 0x6c8a2b0f, 0x6c61a823, 0x6c34ecb5, 0x6c03fc1c, 0x6bced9ff, 0x6b958a54, 0x6b581163, 0x6b1673c1, 0x6ad0b652, 0x6a86de48, 0x6a38f123, 0x69e6f4b1, 0x6990ef0b, 0x6936e697, 0x68d8e206, 0x6876e855, 0x681100c9, 0x67a732f4, 0x673986ac, 0x66c80413, 0x6652b392, 0x65d99dd5, 0x655ccbd3, 0x64dc46c3, 0x64581823, 0x63d049b4, 0x6344e578, 0x62b5f5b2, 0x622384e8, 0x618d9ddc, 0x60f44b91, 0x60579947, 0x5fb79278, 0x5f1442dc, 0x5e6db665, 0x5dc3f93c, 0x5d1717c4, 0x5c671e96, 0x5bb41a80, 0x5afe1886, 0x5a4525df, 0x59894ff3, 0x58caa45b, 0x580930e1, 0x5745037c, 0x567e2a51, 0x55b4b3af, 0x54e8ae13, 0x541a281e, 0x5349309e, 0x5275d684, 0x51a028e8, 0x50c83704, 0x4fee1037, 0x4f11c3fe, 0x4e3361f7, 0x4d52f9df, 0x4c709b8e, 0x4b8c56f8, 0x4aa63c2c, 0x49be5b50, 0x48d4c4a2, 0x47e98874, 0x46fcb72d, 0x460e6148, 0x451e9750, 0x442d69de, 0x433ae99c, 0x4247273f, 0x41523389, 0x405c1f43, 0x3f64fb40, 0x3e6cd85b, 0x3d73c772, 0x3c79d968, 0x3b7f1f23, 0x3a83a989, 0x3987897f, 0x388acfe9, 0x378d8da8, 0x368fd397, 0x3591b28b, 0x34933b50, 0x33947eab, 0x32958d55, 0x319677fa, 0x30974f3b, 0x2f9823a8, 0x2e9905c1, 0x2d9a05f4, 0x2c9b349e, 0x2b9ca203, 0x2a9e5e57, 0x29a079b2, 0x28a30416, 0x27a60d6a, 0x26a9a57b, 0x25addbf9, 0x24b2c075, 0x23b86263, 0x22bed116, 0x21c61bc0, 0x20ce516f, 0x1fd7810f, 0x1ee1b965, 0x1ded0911, 0x1cf97e8b, 0x1c072823, 0x1b1613ff, 0x1a26501b, 0x1937ea47, 0x184af025, 0x175f6f2b, 0x1675749e, 0x158d0d95, 0x14a646f6, 0x13c12d73, 0x12ddcd8f, 0x11fc3395, + 0x111c6ba0, 0x103e8192, 0x0f62811a, 0x0e8875ad, 0x0db06a89, 0x0cda6ab5, 0x0c0680fe, 0x0b34b7f5, 0x0a6519f4, 0x0997b116, 0x08cc873c, 0x0803a60a, 0x073d16e7, 0x0678e2fc, 0x05b71332, 0x04f7b037, 0x043ac276, 0x0380521c, 0x02c86715, 0x0213090c, 0x01603f6e, 0x00b01162, 0x000285d0, 0xff57a35e, 0xfeaf706f, 0xfe09f323, 0xfd673159, 0xfcc730aa, 0xfc29f670, 0xfb8f87bd, 0xfaf7e963, 0xfa631fef, 0xf9d12fab, 0xf9421c9d, 0xf8b5ea87, 0xf82c9ce7, 0xf7a636fa, 0xf722bbb5, 0xf6a22dcf, 0xf6248fb6, 0xf5a9e398, 0xf5322b61, 0xf4bd68b6, 0xf44b9cfe, 0xf3dcc959, 0xf370eea9, 0xf3080d8c, 0xf2a2265e, 0xf23f393b, 0xf1df45fd, 0xf1824c3e, 0xf1284b58, 0xf0d14267, 0xf07d3043, 0xf02c138a, 0xefddea9a, 0xef92b393, 0xef4a6c58, 0xef051290, 0xeec2a3a3, 0xee831cc3, 0xee467ae1, 0xee0cbab9, 0xedd5d8ca, 0xeda1d15c, 0xed70a07d, 0xed424205, 0xed16b196, 0xecedea99, 0xecc7e845, 0xeca4a59b, 0xec841d68, 0xec664a48, 0xec4b26a2, 0xec32acb0, 0xec1cd677, 0xec099dcf, 0xebf8fc64, 0xebeaebaf, 0xebdf6500, 0xebd6617b, 0xebcfda19, 0xebcbc7a7, 0xebca22cc, 0xebcae405, 0xebce03aa, 0xebd379eb, 0xebdb3ed5, 0xebe54a4f, 0xebf1941f, 0xec0013e8, 0xec10c12c, 0xec23934f, 0xec388194, 0xec4f8322, 0xec688f02, 0xec839c22, 0xeca0a156, 0xecbf9558, 0xece06ecb, 0xed032439, 0xed27ac16, 0xed4dfcc2, 0xed760c88, 0xed9fd1a2, 0xedcb4237, 0xedf8545b, 0xee26fe17, 0xee573562, 0xee88f026, 0xeebc2444, 0xeef0c78d, 0xef26cfca, 0xef5e32bd, 0xef96e61c, 0xefd0df9a, 0xf00c14e1, 0xf0487b98, 0xf0860962, 0xf0c4b3e0, 0xf10470b0, 0xf1453571, 0xf186f7c0, 0xf1c9ad40, 0xf20d4b92, 0xf251c85d, 0xf297194d, 0xf2dd3411, + 0xf3240e61, 0xf36b9dfd, 0xf3b3d8ac, 0xf3fcb43e, 0xf4462690, 0xf4902587, 0xf4daa718, 0xf525a143, 0xf5710a17, 0xf5bcd7b1, 0xf609003f, 0xf6557a00, 0xf6a23b44, 0xf6ef3a6e, 0xf73c6df4, 0xf789cc61, 0xf7d74c53, 0xf824e480, 0xf8728bb3, 0xf8c038d0, 0xf90de2d1, 0xf95b80cb, 0xf9a909ea, 0xf9f67577, 0xfa43bad2, 0xfa90d17b, 0xfaddb10c, 0xfb2a513b, 0xfb76a9dd, 0xfbc2b2e4, 0xfc0e6461, 0xfc59b685, 0xfca4a19f, 0xfcef1e20, 0xfd392498, 0xfd82adba, 0xfdcbb25a, 0xfe142b6e, 0xfe5c120f, 0xfea35f79, 0xfeea0d0c, 0xff30144a, 0xff756edc, 0xffba168d, 0xfffe054e, 0x00413536, 0x0083a081, 0x00c54190, 0x010612eb, 0x01460f41, 0x01853165, 0x01c37452, 0x0200d32c, 0x023d493c, 0x0278d1f2, 0x02b368e6, 0x02ed09d7, 0x0325b0ad, 0x035d5977, 0x0394006a, 0x03c9a1e5, 0x03fe3a6f, 0x0431c6b5, 0x0464438c, 0x0495adf2, 0x04c6030d, 0x04f54029, 0x052362ba, 0x0550685d, 0x057c4ed4, 0x05a7140b, 0x05d0b612, 0x05f93324, 0x0620899e, 0x0646b808, 0x066bbd0d, 0x068f9781, 0x06b2465b, 0x06d3c8bb, 0x06f41de3, 0x0713453d, 0x07313e56, 0x074e08e0, 0x0769a4b2, 0x078411c7, 0x079d503b, 0x07b56051, 0x07cc426c, 0x07e1f712, 0x07f67eec, 0x0809dac3, 0x081c0b84, 0x082d1239, 0x083cf010, 0x084ba654, 0x08593671, 0x0865a1f1, 0x0870ea7e, 0x087b11de, 0x088419f6, 0x088c04c8, 0x0892d470, 0x08988b2a, 0x089d2b4a, 0x08a0b740, 0x08a33196, 0x08a49cf0, 0x08a4fc0d, 0x08a451c0, 0x08a2a0f8, 0x089fecbb, 0x089c3824, 0x08978666, 0x0891dac8, 0x088b38a9, 0x0883a378, 0x087b1ebc, 0x0871ae0d, 0x08675516, 0x085c1794, 0x084ff957, 0x0842fe3d, 0x08352a35, 0x0826813e, 0x08170767, 0x0806c0cb, 0x07f5b193, 0x07e3ddf7, + 0x07d14a38, 0x07bdfaa5, 0x07a9f399, 0x07953976, 0x077fd0ac, 0x0769bdaf, 0x07530501, 0x073bab28, 0x0723b4b4, 0x070b2639, 0x06f20453, 0x06d853a2, 0x06be18cd, 0x06a3587e, 0x06881761, 0x066c5a27, 0x06502583, 0x06337e2a, 0x061668d2, 0x05f8ea30, 0x05db06fc, 0x05bcc3ed, 0x059e25b5, 0x057f310a, 0x055fea9d, 0x0540571a, 0x05207b2f, 0x05005b82, 0x04dffcb6, 0x04bf6369, 0x049e9433, 0x047d93a8, 0x045c6654, 0x043b10bd, 0x04199760, 0x03f7feb4, 0x03d64b27, 0x03b4811d, 0x0392a4f4, 0x0370bafc, 0x034ec77f, 0x032ccebb, 0x030ad4e1, 0x02e8de19, 0x02c6ee7f, 0x02a50a22, 0x02833506, 0x02617321, 0x023fc85c, 0x021e3891, 0x01fcc78f, 0x01db7914, 0x01ba50d2, 0x0199526b, 0x01788170, 0x0157e166, 0x013775bf, 0x011741df, 0x00f7491a, 0x00d78eb3, 0x00b815da, 0x0098e1b3, 0x0079f54c, 0x005b53a4, 0x003cffa9, 0x001efc35, 0x00014c12, 0xffe3f1f7, 0xffc6f08a, 0xffaa4a5d, 0xff8e01f1, 0xff7219b3, 0xff5693fe, 0xff3b731b, 0xff20b93e, 0xff066889, 0xfeec830d, 0xfed30ac5, 0xfeba0199, 0xfea16960, 0xfe8943dc, 0xfe7192bd, 0xfe5a579d, 0xfe439407, 0xfe2d496f, 0xfe177937, 0xfe0224b0, 0xfded4d13, 0xfdd8f38b, 0xfdc5192d, 0xfdb1befc, 0xfd9ee5e7, 0xfd8c8ecc, 0xfd7aba74, 0xfd696998, 0xfd589cdc, 0xfd4854d3, 0xfd3891fd, 0xfd2954c8, 0xfd1a9d91, 0xfd0c6ca2, 0xfcfec233, 0xfcf19e6b, 0xfce50161, 0xfcd8eb17, 0xfccd5b82, 0xfcc25285, 0xfcb7cff0, 0xfcadd386, 0xfca45cf7, 0xfc9b6be5, 0xfc92ffe1, 0xfc8b186d, 0xfc83b4fc, 0xfc7cd4f0, 0xfc76779e, 0xfc709c4d, 0xfc6b4233, 0xfc66687a, 0xfc620e3d, 0xfc5e328c, 0xfc5ad465, 0xfc57f2be, 0xfc558c7c, 0xfc53a07b, 0xfc522d88, 0xfc513266, 0xfc50adcc, + 0xfc509e64, 0xfc5102d0, 0xfc51d9a6, 0xfc53216f, 0xfc54d8ae, 0xfc56fdda, 0xfc598f60, 0xfc5c8ba5, 0xfc5ff105, 0xfc63bdd3, 0xfc67f05a, 0xfc6c86dd, 0xfc717f97, 0xfc76d8bc, 0xfc7c9079, 0xfc82a4f4, 0xfc89144d, 0xfc8fdc9f, 0xfc96fbfc, 0xfc9e7074, 0xfca63810, 0xfcae50d6, 0xfcb6b8c4, 0xfcbf6dd8, 0xfcc86e09, 0xfcd1b74c, 0xfcdb4793, 0xfce51ccb, 0xfcef34e1, 0xfcf98dbe, 0xfd04254a, 0xfd0ef969, 0xfd1a0801, 0xfd254ef4, 0xfd30cc24, 0xfd3c7d73, 0xfd4860c2, 0xfd5473f3, 0xfd60b4e7, 0xfd6d2180, 0xfd79b7a1, 0xfd86752e, 0xfd93580d, 0xfda05e23, 0xfdad855b, 0xfdbacb9e, 0xfdc82edb, 0xfdd5ad01, 0xfde34403, 0xfdf0f1d6, 0xfdfeb475, 0xfe0c89db, 0xfe1a7009, 0xfe286505, 0xfe3666d5, 0xfe447389, 0xfe528931, 0xfe60a5e5, 0xfe6ec7c0, 0xfe7cece2, 0xfe8b1373, 0xfe99399f, 0xfea75d97, 0xfeb57d92, 0xfec397cf, 0xfed1aa92, 0xfedfb425, 0xfeedb2da, 0xfefba508, 0xff09890f, 0xff175d53, 0xff252042, 0xff32d04f, 0xff406bf8, 0xff4df1be, 0xff5b602c, 0xff68b5d5, 0xff75f153, 0xff831148, 0xff90145e, 0xff9cf947, 0xffa9bebe, 0xffb66386, 0xffc2e669, 0xffcf463a, 0xffdb81d6, 0xffe79820, 0xfff38806, 0xffff507b, 0x000af07f, 0x00166718, 0x0021b355, 0x002cd44d, 0x0037c922, 0x004290fc, 0x004d2b0e, 0x00579691, 0x0061d2ca, 0x006bdf05, 0x0075ba95, 0x007f64da, 0x0088dd38, 0x0092231e, 0x009b3605, 0x00a4156b, 0x00acc0da, 0x00b537e1, 0x00bd7a1c, 0x00c5872a, 0x00cd5eb7, 0x00d50075, 0x00dc6c1e, 0x00e3a175, 0x00eaa045, 0x00f16861, 0x00f7f9a3, 0x00fe53ef, 0x0104772e, 0x010a6353, 0x01101858, 0x0115963d, 0x011add0b, 0x011fecd3, 0x0124c5ab, 0x012967b1, 0x012dd30a, 0x013207e4, 0x01360670, + 0x0139cee9, 0x013d618d, 0x0140bea5, 0x0143e67c, 0x0146d965, 0x014997bb, 0x014c21db, 0x014e782a, 0x01509b14, 0x01528b08, 0x0154487b, 0x0155d3e8, 0x01572dcf, 0x015856b6, 0x01594f25, 0x015a17ab, 0x015ab0db, 0x015b1b4e, 0x015b579e, 0x015b666c, 0x015b485b, 0x015afe14, 0x015a8843, 0x0159e796, 0x01591cc0, 0x01582878, 0x01570b77, 0x0155c678, 0x01545a3c, 0x0152c783, 0x01510f13, 0x014f31b2, 0x014d3029, 0x014b0b45, 0x0148c3d2, 0x01465a9f, 0x0143d07f, 0x01412643, 0x013e5cc0, 0x013b74ca, 0x01386f3a, 0x01354ce7, 0x01320ea9, 0x012eb55a, 0x012b41d3, 0x0127b4f1, 0x01240f8e, 0x01205285, 0x011c7eb2, 0x011894f0, 0x0114961b, 0x0110830f, 0x010c5ca6, 0x010823ba, 0x0103d927, 0x00ff7dc4, 0x00fb126b, 0x00f697f3, 0x00f20f32, 0x00ed78ff, 0x00e8d62d, 0x00e4278f, 0x00df6df7, 0x00daaa34, 0x00d5dd16, 0x00d10769, 0x00cc29f7, 0x00c7458a, 0x00c25ae8, 0x00bd6ad7, 0x00b87619, 0x00b37d70, 0x00ae8198, 0x00a9834e, 0x00a4834c, 0x009f8249, 0x009a80f8, 0x0095800c, 0x00908034, 0x008b821b, 0x0086866b, 0x00818dcb, 0x007c98de, 0x0077a845, 0x0072bc9d, 0x006dd680, 0x0068f687, 0x00641d44, 0x005f4b4a, 0x005a8125, 0x0055bf60, 0x00510682, 0x004c570f, 0x0047b186, 0x00431666, 0x003e8628, 0x003a0141, 0x00358824, 0x00311b41, 0x002cbb03, 0x002867d2, 0x00242213, 0x001fea27, 0x001bc06b, 0x0017a53b, 0x001398ec, 0x000f9bd2, 0x000bae3c, 0x0007d075, 0x000402c8, 0x00004579, 0xfffc98c9, 0xfff8fcf7, 0xfff5723d, 0xfff1f8d2, 0xffee90eb, 0xffeb3ab8, 0xffe7f666, 0xffe4c41e, 0xffe1a408, 0xffde9646, 0xffdb9af8, 0xffd8b23b, 0xffd5dc28, 0xffd318d6, 0xffd06858, 0xffcdcabe, 0xffcb4014, + 0xffc8c866, 0xffc663b9, 0xffc41212, 0xffc1d373, 0xffbfa7d9, 0xffbd8f40, 0xffbb89a1, 0xffb996f3, 0xffb7b728, 0xffb5ea31, 0xffb42ffc, 0xffb28876, 0xffb0f388, 0xffaf7118, 0xffae010b, 0xffaca344, 0xffab57a1, 0xffaa1e02, 0xffa8f641, 0xffa7e039, 0xffa6dbc0, 0xffa5e8ad, 0xffa506d2, 0xffa43603, 0xffa3760e, 0xffa2c6c2, 0xffa227ec, 0xffa19957, 0xffa11acb, 0xffa0ac11, 0xffa04cf0, 0xff9ffd2c, 0xff9fbc89, 0xff9f8ac9, 0xff9f67ae, 0xff9f52f7, 0xff9f4c65, 0xff9f53b4, 0xff9f68a1, 0xff9f8ae9, 0xff9fba47, 0xff9ff674, 0xffa03f2b, 0xffa09425, 0xffa0f519, 0xffa161bf, 0xffa1d9cf, 0xffa25cfe, 0xffa2eb04, 0xffa38395, 0xffa42668, 0xffa4d332, 0xffa589a6, 0xffa6497c, 0xffa71266, 0xffa7e41a, 0xffa8be4c, 0xffa9a0b1, 0xffaa8afe, 0xffab7ce7, 0xffac7621, 0xffad7662, 0xffae7d5f, 0xffaf8acd, 0xffb09e63, 0xffb1b7d8, 0xffb2d6e1, 0xffb3fb37, 0xffb52490, 0xffb652a7, 0xffb78533, 0xffb8bbed, 0xffb9f691, 0xffbb34d8, 0xffbc767f, 0xffbdbb42, 0xffbf02dd, 0xffc04d0f, 0xffc19996, 0xffc2e832, 0xffc438a3, 0xffc58aaa, 0xffc6de09, 0xffc83285, 0xffc987e0, 0xffcadde1, 0xffcc344c, 0xffcd8aeb, 0xffcee183, 0xffd037e0, 0xffd18dcc, 0xffd2e311, 0xffd4377d, 0xffd58ade, 0xffd6dd02, 0xffd82dba, 0xffd97cd6, 0xffdaca2a, 0xffdc1588, 0xffdd5ec6, 0xffdea5bb, 0xffdfea3c, 0xffe12c22, 0xffe26b48, 0xffe3a788, 0xffe4e0bf, 0xffe616c8, 0xffe74984, 0xffe878d3, 0xffe9a494, 0xffeaccaa, 0xffebf0fa, 0xffed1166, 0xffee2dd7, 0xffef4632, 0xfff05a60, 0xfff16a4a, 0xfff275db, 0xfff37d00, 0xfff47fa5, 0xfff57db8, 0xfff67729, 0xfff76be9, 0xfff85be8, 0xfff9471b, 0xfffa2d74, 0xfffb0ee9, 0xfffbeb70, + 0xfffcc300, 0xfffd9592, 0xfffe631e, 0xffff2b9f, 0xffffef10, 0x0000ad6e, 0x000166b6, 0x00021ae5, 0x0002c9fd, 0x000373fb, 0x000418e2, 0x0004b8b3, 0x00055371, 0x0005e921, 0x000679c5, 0x00070564, 0x00078c04, 0x00080dab, 0x00088a62, 0x00090230, 0x0009751e, 0x0009e337, 0x000a4c85, 0x000ab112, 0x000b10ec, 0x000b6c1d, 0x000bc2b3, 0x000c14bb, 0x000c6244, 0x000cab5c, 0x000cf012, 0x000d3075, 0x000d6c97, 0x000da486, 0x000dd854, 0x000e0812, 0x000e33d3, 0x000e5ba7, 0x000e7fa1, 0x000e9fd5, 0x000ebc54, 0x000ed533, 0x000eea84, 0x000efc5c, 0x000f0ace, 0x000f15ef, 0x000f1dd2, 0x000f228d, 0x000f2434, 0x000f22dc, 0x000f1e99, 0x000f1781, 0x000f0da8, 0x000f0125, 0x000ef20b, 0x000ee070, 0x000ecc69, 0x000eb60b, 0x000e9d6b, 0x000e829e, 0x000e65ba, 0x000e46d3, 0x000e25fd, 0x000e034f, 0x000ddedb, 0x000db8b7, 0x000d90f6, 0x000d67ae, 0x000d3cf1, 0x000d10d5, 0x000ce36b, 0x000cb4c8, 0x000c84ff, 0x000c5422, 0x000c2245, 0x000bef79, 0x000bbbd2, 0x000b8760, 0x000b5235, 0x000b1c64, 0x000ae5fc, 0x000aaf0f, 0x000a77ac, 0x000a3fe5, 0x000a07c9, 0x0009cf67, 0x000996ce, 0x00095e0e, 0x00092535, 0x0008ec50, 0x0008b36e, 0x00087a9c, 0x000841e8, 0x0008095d, 0x0007d108, 0x000798f5, 0x00076130, 0x000729c4, 0x0006f2bb, 0x0006bc21, 0x000685ff, 0x0006505f, 0x00061b4b, 0x0005e6cb, 0x0005b2e8, 0x00057faa, 0x00054d1a, 0x00051b3e, 0x0004ea1d, 0x0004b9c0, 0x00048a2b, 0x00045b65, 0x00042d74, 0x0004005e, 0x0003d426, 0x0003a8d2, 0x00037e65, 0x000354e5, 0x00032c54, 0x000304b7, 0x0002de0e, 0x0002b85f, 0x000293aa, 0x00026ff2, 0x00024d39, 0x00022b7f, 0x00020ac7, 0x0001eb10, 0x00000000 // this one is needed for lerping the last coefficient }; /* - * 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 16600 */ 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, + 0x58888889, 0x58875d88, 0x5883dc96, 0x587e05e0, 0x5875d9b3, 0x586b587d, 0x585e82c6, 0x584f593a, 0x583ddc9f, 0x582a0dde, 0x5813edfb, 0x57fb7e1a, 0x57e0bf7f, 0x57c3b389, 0x57a45bb8, 0x5782b9aa, 0x575ecf1a, 0x57389de0, 0x571027f6, 0x56e56f6f, 0x56b8767e, 0x56893f73, 0x5657ccbb, 0x562420e2, 0x55ee3e8d, 0x55b62882, 0x557be1a0, 0x553f6ce6, 0x5500cd6d, 0x54c0066a, 0x547d1b2e, 0x54380f26, 0x53f0e5da, 0x53a7a2ed, 0x535c4a1e, 0x530edf46, 0x52bf6657, 0x526de360, 0x521a5a86, 0x51c4d00c, 0x516d484a, 0x5113c7b6, 0x50b852d9, 0x505aee59, 0x4ffb9ef2, 0x4f9a6979, 0x4f3752d9, 0x4ed26016, 0x4e6b9649, 0x4e02faa3, 0x4d98926b, 0x4d2c62fd, 0x4cbe71cc, 0x4c4ec45e, 0x4bdd6050, 0x4b6a4b53, 0x4af58b2b, 0x4a7f25b0, 0x4a0720cd, 0x498d8283, 0x491250e1, 0x4895920c, 0x48174c37, 0x479785ab, 0x471644bd, 0x46938fd7, 0x460f6d70, 0x4589e411, 0x4502fa51, 0x447ab6d5, 0x43f12053, 0x43663d8d, 0x42da1554, 0x424cae85, 0x41be100a, 0x412e40db, 0x409d47f9, 0x400b2c72, 0x3f77f561, 0x3ee3a9e7, 0x3e4e5132, 0x3db7f27a, 0x3d2094ff, 0x3c88400b, 0x3beefaee, 0x3b54cd01, 0x3ab9bda6, 0x3a1dd444, 0x39811848, 0x38e39127, 0x3845465a, 0x37a63f5f, 0x370683ba, 0x36661af1, 0x35c50c90, 0x35236024, 0x34811d3f, 0x33de4b72, 0x333af253, 0x32971979, 0x31f2c87a, 0x314e06ed, 0x30a8dc6a, 0x30035089, 0x2f5d6ade, 0x2eb732fe, 0x2e10b07d, 0x2d69eaeb, 0x2cc2e9d4, 0x2c1bb4c4, 0x2b745340, 0x2acccccc, 0x2a2528e6, 0x297d6f06, 0x28d5a6a0, 0x282dd722, 0x278607f2, 0x26de4072, 0x263687fa, 0x258ee5dd, 0x24e76163, 0x244001cf, 0x2398ce58, 0x22f1ce2e, 0x224b0876, 0x21a4844b, 0x20fe48be, 0x20585cd5, + 0x1fb2c78a, 0x1f0d8fcb, 0x1e68bc7d, 0x1dc45475, 0x1d205e7d, 0x1c7ce150, 0x1bd9e39e, 0x1b376c06, 0x1a95811c, 0x19f42964, 0x19536b51, 0x18b34d4a, 0x1813d5a3, 0x17750aa3, 0x16d6f27f, 0x1639935b, 0x159cf34b, 0x15011851, 0x1466085d, 0x13cbc94f, 0x133260f3, 0x1299d502, 0x12022b24, 0x116b68ed, 0x10d593dd, 0x1040b162, 0x0facc6d4, 0x0f19d979, 0x0e87ee81, 0x0df70b09, 0x0d673417, 0x0cd86e9d, 0x0c4abf78, 0x0bbe2b70, 0x0b32b735, 0x0aa86763, 0x0a1f407f, 0x099746f9, 0x09107f29, 0x088aed4f, 0x08069598, 0x07837c17, 0x0701a4c8, 0x06811392, 0x0601cc40, 0x0583d28b, 0x05072a0f, 0x048bd653, 0x0411dac7, 0x03993abf, 0x0321f97b, 0x02ac1a20, 0x02379fbb, 0x01c48d42, 0x0152e590, 0x00e2ab69, 0x0073e179, 0x00068a52, 0xff9aa86c, 0xff303e29, 0xfec74dd1, 0xfe5fd993, 0xfdf9e383, 0xfd956da0, 0xfd3279cd, 0xfcd109d6, 0xfc711f6d, 0xfc12bc2a, 0xfbb5e18f, 0xfb5a9103, 0xfb00cbd4, 0xfaa89339, 0xfa51e84e, 0xf9fccc18, 0xf9a93f82, 0xf9574361, 0xf906d86d, 0xf8b7ff4b, 0xf86ab883, 0xf81f0487, 0xf7d4e3b0, 0xf78c5641, 0xf7455c62, 0xf6fff625, 0xf6bc2385, 0xf679e463, 0xf639388a, 0xf5fa1fae, 0xf5bc996b, 0xf580a547, 0xf54642b1, 0xf50d70ff, 0xf4d62f74, 0xf4a07d3b, 0xf46c5967, 0xf439c2f9, 0xf408b8d8, 0xf3d939d9, 0xf3ab44b9, 0xf37ed821, 0xf353f2a5, 0xf32a92c3, 0xf302b6e6, 0xf2dc5d64, 0xf2b7847f, 0xf2942a64, 0xf2724d2e, 0xf251eae4, 0xf2330179, 0xf2158ece, 0xf1f990b1, 0xf1df04de, 0xf1c5e8ff, 0xf1ae3aaa, 0xf197f765, 0xf1831ca6, 0xf16fa7d0, 0xf15d9634, 0xf14ce516, 0xf13d91a7, 0xf12f9909, 0xf122f84e, 0xf117ac79, 0xf10db27d, 0xf1050741, 0xf0fda799, 0xf0f7904e, 0xf0f2be1a, + 0xf0ef2dab, 0xf0ecdba0, 0xf0ebc48a, 0xf0ebe4f1, 0xf0ed394e, 0xf0efbe0d, 0xf0f36f92, 0xf0f84a32, 0xf0fe4a39, 0xf1056be8, 0xf10dab74, 0xf117050a, 0xf12174cd, 0xf12cf6d5, 0xf1398732, 0xf14721ec, 0xf155c300, 0xf1656666, 0xf176080d, 0xf187a3db, 0xf19a35b1, 0xf1adb969, 0xf1c22ad4, 0xf1d785c1, 0xf1edc5f5, 0xf204e733, 0xf21ce537, 0xf235bbb8, 0xf24f6669, 0xf269e0fa, 0xf2852715, 0xf2a13462, 0xf2be0485, 0xf2db9321, 0xf2f9dbd3, 0xf318da38, 0xf33889ec, 0xf358e688, 0xf379eba4, 0xf39b94d7, 0xf3bdddb7, 0xf3e0c1db, 0xf4043cd8, 0xf4284a45, 0xf44ce5ba, 0xf4720ace, 0xf497b51a, 0xf4bde03a, 0xf4e487c9, 0xf50ba766, 0xf5333ab3, 0xf55b3d52, 0xf583aaec, 0xf5ac7f29, 0xf5d5b5b7, 0xf5ff4a47, 0xf6293890, 0xf6537c4a, 0xf67e1134, 0xf6a8f311, 0xf6d41dab, 0xf6ff8cce, 0xf72b3c4f, 0xf7572808, 0xf7834bd7, 0xf7afa3a3, 0xf7dc2b58, 0xf808deec, 0xf835ba59, 0xf862b9a0, 0xf88fd8cc, 0xf8bd13f0, 0xf8ea6724, 0xf917ce8a, 0xf945464f, 0xf972caa4, 0xf9a057c6, 0xf9cde9fb, 0xf9fb7d90, 0xfa290edf, 0xfa569a49, 0xfa841c3a, 0xfab19127, 0xfadef591, 0xfb0c4601, 0xfb397f0d, 0xfb669d55, 0xfb939d83, 0xfbc07c4c, 0xfbed3671, 0xfc19c8bf, 0xfc46300d, 0xfc72693e, 0xfc9e7141, 0xfcca4511, 0xfcf5e1b4, 0xfd21443e, 0xfd4c69cd, 0xfd774f8e, 0xfda1f2b7, 0xfdcc508d, 0xfdf66662, 0xfe203193, 0xfe49af8a, 0xfe72ddbf, 0xfe9bb9b7, 0xfec44103, 0xfeec7141, 0xff14481d, 0xff3bc351, 0xff62e0a2, 0xff899de5, 0xffaff8f9, 0xffd5efce, 0xfffb8060, 0x0020a8b7, 0x004566eb, 0x0069b920, 0x008d9d89, 0x00b11264, 0x00d415ff, 0x00f6a6b5, 0x0118c2ef, 0x013a6922, 0x015b97d1, 0x017c4d8f, 0x019c88f9, 0x01bc48bd, + 0x01db8b94, 0x01fa5045, 0x021895a6, 0x02365a98, 0x02539e0b, 0x02705efd, 0x028c9c77, 0x02a85592, 0x02c38972, 0x02de3749, 0x02f85e57, 0x0311fde7, 0x032b1552, 0x0343a3ff, 0x035ba961, 0x037324f6, 0x038a164c, 0x03a07cfa, 0x03b658a7, 0x03cba904, 0x03e06dcf, 0x03f4a6d1, 0x040853e2, 0x041b74e4, 0x042e09c4, 0x0440127d, 0x04518f14, 0x04627f9b, 0x0472e42e, 0x0482bcf5, 0x04920a24, 0x04a0cbf7, 0x04af02ba, 0x04bcaebe, 0x04c9d064, 0x04d66814, 0x04e27642, 0x04edfb6c, 0x04f8f819, 0x05036cdc, 0x050d5a51, 0x0516c11c, 0x051fa1ee, 0x0527fd7e, 0x052fd48d, 0x053727e8, 0x053df861, 0x054446d5, 0x054a1429, 0x054f614a, 0x05542f2f, 0x05587ed5, 0x055c5141, 0x055fa783, 0x056282ae, 0x0564e3e1, 0x0566cc3e, 0x05683cf1, 0x0569372c, 0x0569bc29, 0x0569cd27, 0x05696b6b, 0x05689842, 0x056754fe, 0x0565a2f9, 0x0563838f, 0x0560f824, 0x055e0222, 0x055aa2f6, 0x0556dc14, 0x0552aef5, 0x054e1d14, 0x054927f4, 0x0543d11a, 0x053e1a11, 0x05380465, 0x053191aa, 0x052ac373, 0x05239b5b, 0x051c1afe, 0x051443fa, 0x050c17f3, 0x0503988d, 0x04fac770, 0x04f1a647, 0x04e836bd, 0x04de7a82, 0x04d47346, 0x04ca22bc, 0x04bf8a97, 0x04b4ac8c, 0x04a98a54, 0x049e25a4, 0x04928037, 0x04869bc6, 0x047a7a0b, 0x046e1cc1, 0x046185a3, 0x0454b66c, 0x0447b0d7, 0x043a76a1, 0x042d0983, 0x041f6b39, 0x04119d7b, 0x0403a204, 0x03f57a8c, 0x03e728c9, 0x03d8ae73, 0x03ca0d3e, 0x03bb46dd, 0x03ac5d03, 0x039d5160, 0x038e25a2, 0x037edb76, 0x036f7486, 0x035ff27a, 0x035056f9, 0x0340a3a5, 0x0330da20, 0x0320fc08, 0x03110af8, 0x03010889, 0x02f0f64f, 0x02e0d5df, 0x02d0a8c6, 0x02c07090, 0x02b02ec6, 0x029fe4ec, + 0x028f9484, 0x027f3f0b, 0x026ee5fa, 0x025e8ac8, 0x024e2ee5, 0x023dd3c0, 0x022d7ac1, 0x021d254d, 0x020cd4c6, 0x01fc8a88, 0x01ec47ea, 0x01dc0e40, 0x01cbded8, 0x01bbbafd, 0x01aba3f2, 0x019b9afa, 0x018ba14e, 0x017bb826, 0x016be0b3, 0x015c1c20, 0x014c6b97, 0x013cd038, 0x012d4b20, 0x011ddd67, 0x010e8820, 0x00ff4c57, 0x00f02b13, 0x00e12558, 0x00d23c22, 0x00c37068, 0x00b4c31c, 0x00a6352a, 0x0097c778, 0x00897ae9, 0x007b5057, 0x006d4899, 0x005f647f, 0x0051a4d3, 0x00440a5a, 0x003695d5, 0x002947fc, 0x001c2183, 0x000f231a, 0x00024d68, 0xfff5a111, 0xffe91eb2, 0xffdcc6e4, 0xffd09a37, 0xffc49939, 0xffb8c471, 0xffad1c5f, 0xffa1a180, 0xff965449, 0xff8b352a, 0xff804490, 0xff7582e0, 0xff6af079, 0xff608db6, 0xff565aec, 0xff4c586c, 0xff42867e, 0xff38e569, 0xff2f756c, 0xff2636c2, 0xff1d29a0, 0xff144e36, 0xff0ba4ae, 0xff032d30, 0xfefae7db, 0xfef2d4cc, 0xfeeaf419, 0xfee345d5, 0xfedbca0b, 0xfed480c6, 0xfecd6a07, 0xfec685cf, 0xfebfd416, 0xfeb954d4, 0xfeb307f8, 0xfeaced6f, 0xfea70522, 0xfea14ef4, 0xfe9bcac5, 0xfe96786f, 0xfe9157cb, 0xfe8c68ab, 0xfe87aadd, 0xfe831e2e, 0xfe7ec263, 0xfe7a9741, 0xfe769c85, 0xfe72d1ed, 0xfe6f3731, 0xfe6bcc04, 0xfe689017, 0xfe658319, 0xfe62a4b3, 0xfe5ff48d, 0xfe5d7249, 0xfe5b1d89, 0xfe58f5ea, 0xfe56fb06, 0xfe552c76, 0xfe5389cc, 0xfe52129d, 0xfe50c676, 0xfe4fa4e5, 0xfe4ead73, 0xfe4ddfa8, 0xfe4d3b09, 0xfe4cbf19, 0xfe4c6b59, 0xfe4c3f47, 0xfe4c3a5e, 0xfe4c5c1b, 0xfe4ca3f4, 0xfe4d1160, 0xfe4da3d4, 0xfe4e5ac3, 0xfe4f359e, 0xfe5033d5, 0xfe5154d6, 0xfe52980d, 0xfe53fce6, 0xfe5582cb, 0xfe572926, 0xfe58ef5d, 0xfe5ad4d7, + 0xfe5cd8fa, 0xfe5efb2b, 0xfe613ace, 0xfe639746, 0xfe660ff5, 0xfe68a43c, 0xfe6b537e, 0xfe6e1d1b, 0xfe710072, 0xfe73fce5, 0xfe7711d2, 0xfe7a3e98, 0xfe7d8297, 0xfe80dd2e, 0xfe844dbc, 0xfe87d39f, 0xfe8b6e37, 0xfe8f1ce3, 0xfe92df02, 0xfe96b3f4, 0xfe9a9b19, 0xfe9e93d1, 0xfea29d7d, 0xfea6b77d, 0xfeaae135, 0xfeaf1a05, 0xfeb36152, 0xfeb7b67e, 0xfebc18ef, 0xfec0880a, 0xfec50334, 0xfec989d5, 0xfece1b54, 0xfed2b71b, 0xfed75c94, 0xfedc0b2a, 0xfee0c249, 0xfee5815e, 0xfeea47d8, 0xfeef1528, 0xfef3e8be, 0xfef8c20c, 0xfefda088, 0xff0283a5, 0xff076adc, 0xff0c55a4, 0xff114377, 0xff1633d0, 0xff1b262d, 0xff201a0c, 0xff250eee, 0xff2a0453, 0xff2ef9c1, 0xff33eebc, 0xff38e2cb, 0xff3dd578, 0xff42c64c, 0xff47b4d6, 0xff4ca0a2, 0xff518941, 0xff566e47, 0xff5b4f45, 0xff602bd4, 0xff65038a, 0xff69d601, 0xff6ea2d6, 0xff7369a7, 0xff782a12, 0xff7ce3bb, 0xff819645, 0xff864157, 0xff8ae498, 0xff8f7fb2, 0xff941251, 0xff989c25, 0xff9d1cdc, 0xffa1942a, 0xffa601c3, 0xffaa655e, 0xffaebeb2, 0xffb30d7c, 0xffb75177, 0xffbb8a62, 0xffbfb7ff, 0xffc3da11, 0xffc7f05c, 0xffcbfaa8, 0xffcff8be, 0xffd3ea6a, 0xffd7cf79, 0xffdba7b9, 0xffdf72fe, 0xffe33119, 0xffe6e1e1, 0xffea852e, 0xffee1ad8, 0xfff1a2bb, 0xfff51cb5, 0xfff888a4, 0xfffbe66b, 0xffff35ed, 0x0002770f, 0x0005a9b8, 0x0008cdd0, 0x000be344, 0x000ee9ff, 0x0011e1f0, 0x0014cb08, 0x0017a538, 0x001a7075, 0x001d2cb3, 0x001fd9eb, 0x00227816, 0x0025072f, 0x00278731, 0x0029f81b, 0x002c59ed, 0x002eaca8, 0x0030f04f, 0x003324e6, 0x00354a74, 0x003760ff, 0x00396892, 0x003b6135, 0x003d4af6, 0x003f25e1, 0x0040f206, 0x0042af73, + 0x00445e3a, 0x0045fe6e, 0x00479023, 0x0049136d, 0x004a8864, 0x004bef1e, 0x004d47b5, 0x004e9242, 0x004fcedf, 0x0050fdaa, 0x00521ebe, 0x0053323b, 0x0054383e, 0x005530e9, 0x00561c5b, 0x0056fab7, 0x0057cc20, 0x005890b9, 0x005948a7, 0x0059f40e, 0x005a9315, 0x005b25e2, 0x005bac9d, 0x005c276d, 0x005c967d, 0x005cf9f4, 0x005d51fd, 0x005d9ec3, 0x005de071, 0x005e1731, 0x005e4331, 0x005e649d, 0x005e7ba1, 0x005e886c, 0x005e8b2b, 0x005e840c, 0x005e733e, 0x005e58ef, 0x005e354e, 0x005e088c, 0x005dd2d6, 0x005d945e, 0x005d4d53, 0x005cfde5, 0x005ca645, 0x005c46a2, 0x005bdf2d, 0x005b7017, 0x005af990, 0x005a7bc9, 0x0059f6f2, 0x00596b3b, 0x0058d8d6, 0x00583ff2, 0x0057a0c0, 0x0056fb70, 0x00565032, 0x00559f36, 0x0054e8ac, 0x00542cc2, 0x00536baa, 0x0052a591, 0x0051daa6, 0x00510b19, 0x00503717, 0x004f5ece, 0x004e826d, 0x004da220, 0x004cbe15, 0x004bd678, 0x004aeb75, 0x0049fd39, 0x00490bef, 0x004817c2, 0x004720dd, 0x0046276a, 0x00452b92, 0x00442d80, 0x00432d5b, 0x00422b4c, 0x0041277c, 0x00402210, 0x003f1b31, 0x003e1304, 0x003d09b0, 0x003bff58, 0x003af423, 0x0039e833, 0x0038dbad, 0x0037ceb3, 0x0036c168, 0x0035b3ed, 0x0034a664, 0x003398ed, 0x00328ba7, 0x00317eb3, 0x0030722e, 0x002f6638, 0x002e5aec, 0x002d5069, 0x002c46c9, 0x002b3e2a, 0x002a36a5, 0x00293054, 0x00282b52, 0x002727b7, 0x0026259c, 0x00252518, 0x00242641, 0x00232930, 0x00222df8, 0x002134b0, 0x00203d6b, 0x001f483d, 0x001e5539, 0x001d6473, 0x001c75fb, 0x001b89e3, 0x001aa03b, 0x0019b913, 0x0018d47b, 0x0017f281, 0x00171334, 0x001636a0, 0x00155cd2, 0x001485d7, 0x0013b1ba, 0x0012e086, + 0x00121246, 0x00114703, 0x00107ec6, 0x000fb999, 0x000ef783, 0x000e388c, 0x000d7cba, 0x000cc414, 0x000c0ea0, 0x000b5c64, 0x000aad63, 0x000a01a2, 0x00095925, 0x0008b3f0, 0x00081204, 0x00077364, 0x0006d811, 0x0006400e, 0x0005ab5a, 0x000519f6, 0x00048be2, 0x0004011d, 0x000379a7, 0x0002f57d, 0x0002749e, 0x0001f708, 0x00017cb7, 0x000105a9, 0x000091da, 0x00002147, 0xffffb3eb, 0xffff49c1, 0xfffee2c6, 0xfffe7ef2, 0xfffe1e41, 0xfffdc0ad, 0xfffd6630, 0xfffd0ec3, 0xfffcba5f, 0xfffc68fd, 0xfffc1a97, 0xfffbcf23, 0xfffb869a, 0xfffb40f4, 0xfffafe29, 0xfffabe30, 0xfffa8100, 0xfffa4690, 0xfffa0ed7, 0xfff9d9cc, 0xfff9a764, 0xfff97796, 0xfff94a58, 0xfff91fa0, 0xfff8f764, 0xfff8d199, 0xfff8ae34, 0xfff88d2b, 0xfff86e74, 0xfff85203, 0xfff837cd, 0xfff81fc7, 0xfff809e6, 0xfff7f61f, 0xfff7e467, 0xfff7d4b1, 0xfff7c6f4, 0xfff7bb22, 0xfff7b132, 0xfff7a917, 0xfff7a2c6, 0xfff79e33, 0xfff79b52, 0xfff79a19, 0xfff79a7b, 0xfff79c6e, 0xfff79fe5, 0xfff7a4d5, 0xfff7ab33, 0xfff7b2f3, 0xfff7bc0a, 0xfff7c66d, 0xfff7d210, 0xfff7dee8, 0xfff7eceb, 0xfff7fc0c, 0xfff80c41, 0xfff81d80, 0xfff82fbc, 0xfff842ed, 0xfff85707, 0xfff86bff, 0xfff881cb, 0xfff89861, 0xfff8afb7, 0xfff8c7c3, 0xfff8e07b, 0xfff8f9d4, 0xfff913c6, 0xfff92e46, 0xfff9494c, 0xfff964ce, 0xfff980c3, 0xfff99d23, 0xfff9b9e3, 0xfff9d6fc, 0xfff9f465, 0xfffa1216, 0xfffa3006, 0xfffa4e2d, 0xfffa6c84, 0xfffa8b03, 0xfffaa9a3, 0xfffac85b, 0xfffae725, 0xfffb05f9, 0xfffb24d2, 0xfffb43a7, 0xfffb6273, 0xfffb812f, 0xfffb9fd5, 0xfffbbe5f, 0xfffbdcc6, 0xfffbfb07, 0xfffc191a, 0xfffc36fa, 0xfffc54a4, 0xfffc7210, 0x00000000 // this one is needed for lerping the last coefficient }; @@ -144,11 +141,8 @@ 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 } @@ -163,9 +157,7 @@ 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 } @@ -187,13 +179,8 @@ 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 } diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h index 25fc025..48bc747 100644 --- a/services/audioflinger/AudioResamplerSinc.h +++ b/services/audioflinger/AudioResamplerSinc.h @@ -71,7 +71,7 @@ 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 diff --git a/services/audioflinger/StateQueue.h b/services/audioflinger/StateQueue.h index eba190c..c9b5111 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 diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp new file mode 100644 index 0000000..151313b --- /dev/null +++ b/services/audioflinger/test-resample.cpp @@ -0,0 +1,233 @@ +/* + * 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 "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> + +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] [-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,"-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[]) { + + bool profiling = false; + bool writeHeader = false; + int input_freq = 0; + int output_freq = 0; + AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; + + int ch; + while ((ch = getopt(argc, argv, "phq:i:o:")) != -1) { + switch (ch) { + case 'p': + profiling = true; + break; + case 'h': + writeHeader = true; + 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(argv[0]); + return -1; + } + break; + case 'i': + input_freq = atoi(optarg); + break; + case 'o': + output_freq = atoi(optarg); + break; + case '?': + default: + usage(argv[0]); + return -1; + } + } + argc -= optind; + + if (argc != 2) { + usage(argv[0]); + return -1; + } + + argv += optind; + + // ---------------------------------------------------------- + + struct stat st; + if (stat(argv[0], &st) < 0) { + fprintf(stderr, "stat: %s\n", strerror(errno)); + return -1; + } + + int input_fd = open(argv[0], O_RDONLY); + if (input_fd < 0) { + fprintf(stderr, "open: %s\n", strerror(errno)); + return -1; + } + + size_t input_size = st.st_size; + void* 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; + } + +// printf("input sample rate: %d Hz\n", input_freq); +// printf("output sample rate: %d Hz\n", output_freq); +// printf("input mmap: %p, size=%u\n", input_vaddr, input_size); + + // ---------------------------------------------------------- + + class Provider: public AudioBufferProvider { + int16_t* mAddr; + size_t mNumFrames; + public: + Provider(const void* addr, size_t size) { + mAddr = (int16_t*) addr; + mNumFrames = size / 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); + + size_t output_size = 2 * 2 * ((int64_t) input_size * output_freq) + / input_freq; + output_size &= ~7; // always stereo, 32-bits + + void* output_vaddr = malloc(output_size); + memset(output_vaddr, 0, output_size); + + AudioResampler* resampler = AudioResampler::create(16, 1, output_freq, + quality); + + size_t out_frames = output_size/8; + resampler->setSampleRate(input_freq); + resampler->setVolume(0x1000, 0x1000); + resampler->resample((int*) output_vaddr, out_frames, &provider); + + if (profiling) { + memset(output_vaddr, 0, output_size); + timespec start, end; + clock_gettime(CLOCK_MONOTONIC_HR, &start); + 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; + printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6); + } + + // 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 * sizeof(int16_t)); + for (size_t i = 0; i < out_frames; i++) { + int32_t s = out[i * 2] >> 12; + if (s > 32767) s = 32767; + else if (s < -32768) s = -32768; + convert[i] = int16_t(s); + } + + // write output to disk + int output_fd = open(argv[1], 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*sizeof(int16_t), 1, output_freq, 16); + write(output_fd, &wav, sizeof(wav)); + } + + write(output_fd, convert, out_frames * 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..ea3ef50 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 +121,162 @@ 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 + // total number of coefficients (one side) const int N = (1 << nz) * 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<N ; i++) { + double x = (2.0 * M_PI * i * Fcr) / (1 << nz); + double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr; + y *= atten; + + if (!debug) { + if ((i % (1<<nz)) == 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) { + if (!format) { + printf("\n 0x%08x ", 0); + } else { + printf("\n %.9g ", 0.0f); + } + 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 - + |