diff options
| -rw-r--r-- | include/media/AudioRecord.h | 14 | ||||
| -rw-r--r-- | include/media/AudioSystem.h | 10 | ||||
| -rw-r--r-- | include/media/IAudioFlinger.h | 12 | ||||
| -rw-r--r-- | include/media/IAudioPolicyService.h | 4 | ||||
| -rw-r--r-- | media/libmedia/AudioRecord.cpp | 12 | ||||
| -rw-r--r-- | media/libmedia/AudioSystem.cpp | 8 | ||||
| -rw-r--r-- | media/libmedia/IAudioFlinger.cpp | 32 | ||||
| -rw-r--r-- | media/libmedia/IAudioPolicyService.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 48 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 32 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.h | 4 |
12 files changed, 96 insertions, 96 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 44925f2..756e91d 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -68,7 +68,7 @@ public: }; uint32_t flags; int channelCount; - int format; + audio_format_t format; size_t frameCount; size_t size; union { @@ -112,7 +112,7 @@ public: static status_t getMinFrameCount(int* frameCount, uint32_t sampleRate, - int format, + audio_format_t format, int channelCount); /* Constructs an uninitialized AudioRecord. No connection with @@ -151,7 +151,7 @@ public: AudioRecord(int inputSource, uint32_t sampleRate = 0, - int format = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channelMask = AUDIO_CHANNEL_IN_MONO, int frameCount = 0, uint32_t flags = 0, @@ -177,7 +177,7 @@ public: * */ status_t set(int inputSource = 0, uint32_t sampleRate = 0, - int format = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channelMask = AUDIO_CHANNEL_IN_MONO, int frameCount = 0, uint32_t flags = 0, @@ -203,7 +203,7 @@ public: /* getters, see constructor */ - int format() const; + audio_format_t format() const; int channelCount() const; int channels() const; uint32_t frameCount() const; @@ -349,7 +349,7 @@ private: bool processAudioBuffer(const sp<ClientRecordThread>& thread); status_t openRecord_l(uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -365,7 +365,7 @@ private: uint32_t mFrameCount; audio_track_cblk_t* mCblk; - uint32_t mFormat; + audio_format_t mFormat; uint8_t mChannelCount; uint8_t mInputSource; uint8_t mReserved[2]; diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 49e5690..c6368fb 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -95,7 +95,7 @@ public: static bool routedToA2dpOutput(audio_stream_type_t streamType); - static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, + static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount, size_t* buffSize); static status_t setVoiceVolume(float volume); @@ -134,7 +134,7 @@ public: class OutputDescriptor { public: OutputDescriptor() - : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} + : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {} uint32_t samplingRate; int32_t format; @@ -153,7 +153,7 @@ public: static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); static audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = AUDIO_CHANNEL_OUT_STEREO, audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT); static status_t startOutput(audio_io_handle_t output, @@ -165,7 +165,7 @@ public: static void releaseOutput(audio_io_handle_t output); static audio_io_handle_t getInput(int inputSource, uint32_t samplingRate = 0, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = AUDIO_CHANNEL_IN_MONO, audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0, int sessionId = 0); @@ -242,7 +242,7 @@ private: static size_t gInBuffSize; // previous parameters for recording buffer size queries static uint32_t gPrevInSamplingRate; - static int gPrevInFormat; + static audio_format_t gPrevInFormat; static int gPrevInChannelCount; static sp<IAudioPolicyService> gAudioPolicyService; diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 3999479..7c0d886 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -48,7 +48,7 @@ public: pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -61,7 +61,7 @@ public: pid_t pid, int input, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -73,7 +73,7 @@ public: */ virtual uint32_t sampleRate(int output) const = 0; virtual int channelCount(int output) const = 0; - virtual uint32_t format(int output) const = 0; + virtual audio_format_t format(int output) const = 0; virtual size_t frameCount(int output) const = 0; virtual uint32_t latency(int output) const = 0; @@ -109,11 +109,11 @@ public: virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0; // retrieve the audio recording buffer size - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0; + virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) = 0; virtual int openOutput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, uint32_t flags) = 0; @@ -124,7 +124,7 @@ public: virtual int openInput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t acoustics) = 0; virtual status_t closeInput(int input) = 0; diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index 8ddbe0a..07d17c5 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -50,7 +50,7 @@ public: 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, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = 0, audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) = 0; virtual status_t startOutput(audio_io_handle_t output, @@ -62,7 +62,7 @@ public: virtual void releaseOutput(audio_io_handle_t output) = 0; virtual audio_io_handle_t getInput(int inputSource, uint32_t samplingRate = 0, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = 0, audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0, int audioSession = 0) = 0; diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 32b5bac..5b5b076 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -48,7 +48,7 @@ namespace android { status_t AudioRecord::getMinFrameCount( int* frameCount, uint32_t sampleRate, - int format, + audio_format_t format, int channelCount) { size_t size = 0; @@ -86,7 +86,7 @@ AudioRecord::AudioRecord() AudioRecord::AudioRecord( int inputSource, uint32_t sampleRate, - int format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -121,7 +121,7 @@ AudioRecord::~AudioRecord() status_t AudioRecord::set( int inputSource, uint32_t sampleRate, - int format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -148,7 +148,7 @@ status_t AudioRecord::set( sampleRate = DEFAULT_SAMPLE_RATE; } // these below should probably come from the audioFlinger too... - if (format == 0) { + if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } // validate parameters @@ -248,7 +248,7 @@ uint32_t AudioRecord::latency() const return mLatency; } -int AudioRecord::format() const +audio_format_t AudioRecord::format() const { return mFormat; } @@ -448,7 +448,7 @@ unsigned int AudioRecord::getInputFramesLost() // must be called with mLock held status_t AudioRecord::openRecord_l( uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 5ca868a..952d634 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -40,7 +40,7 @@ DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSyst // Cached values for recording queries, all protected by gLock uint32_t AudioSystem::gPrevInSamplingRate = 16000; -int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT; +audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT; int AudioSystem::gPrevInChannelCount = 1; size_t AudioSystem::gInBuffSize = 0; @@ -308,7 +308,7 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t st return NO_ERROR; } -status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount, +status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount, size_t* buffSize) { gLock.lock(); @@ -572,7 +572,7 @@ audio_policy_forced_cfg_t AudioSystem::getForceUse(audio_policy_force_use_t usag audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_policy_output_flags_t flags) { @@ -632,7 +632,7 @@ void AudioSystem::releaseOutput(audio_io_handle_t output) audio_io_handle_t AudioSystem::getInput(int inputSource, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_in_acoustics_t acoustics, int sessionId) diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index eef551c..0d442ef 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -84,7 +84,7 @@ public: pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -131,7 +131,7 @@ public: pid_t pid, int input, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -188,13 +188,13 @@ public: return reply.readInt32(); } - virtual uint32_t format(int output) const + virtual audio_format_t format(int output) const { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); data.writeInt32(output); remote()->transact(FORMAT, data, &reply); - return reply.readInt32(); + return (audio_format_t) reply.readInt32(); } virtual size_t frameCount(int output) const @@ -343,7 +343,7 @@ public: remote()->transact(REGISTER_CLIENT, data, &reply); } - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) + virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -356,7 +356,7 @@ public: virtual int openOutput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, uint32_t flags) @@ -364,7 +364,7 @@ public: Parcel data, reply; uint32_t devices = pDevices ? *pDevices : 0; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; + audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t channels = pChannels ? *pChannels : 0; uint32_t latency = pLatencyMs ? *pLatencyMs : 0; @@ -382,7 +382,7 @@ public: if (pDevices) *pDevices = devices; samplingRate = reply.readInt32(); if (pSamplingRate) *pSamplingRate = samplingRate; - format = reply.readInt32(); + format = (audio_format_t) reply.readInt32(); if (pFormat) *pFormat = format; channels = reply.readInt32(); if (pChannels) *pChannels = channels; @@ -430,14 +430,14 @@ public: virtual int openInput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t acoustics) { Parcel data, reply; uint32_t devices = pDevices ? *pDevices : 0; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; + audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t channels = pChannels ? *pChannels : 0; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -452,7 +452,7 @@ public: if (pDevices) *pDevices = devices; samplingRate = reply.readInt32(); if (pSamplingRate) *pSamplingRate = samplingRate; - format = reply.readInt32(); + format = (audio_format_t) reply.readInt32(); if (pFormat) *pFormat = format; channels = reply.readInt32(); if (pChannels) *pChannels = channels; @@ -678,7 +678,7 @@ status_t BnAudioFlinger::onTransact( pid_t pid = data.readInt32(); int streamType = data.readInt32(); uint32_t sampleRate = data.readInt32(); - int format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); int channelCount = data.readInt32(); size_t bufferCount = data.readInt32(); uint32_t flags = data.readInt32(); @@ -699,7 +699,7 @@ status_t BnAudioFlinger::onTransact( pid_t pid = data.readInt32(); int input = data.readInt32(); uint32_t sampleRate = data.readInt32(); - int format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); int channelCount = data.readInt32(); size_t bufferCount = data.readInt32(); uint32_t flags = data.readInt32(); @@ -825,7 +825,7 @@ status_t BnAudioFlinger::onTransact( case GET_INPUTBUFFERSIZE: { CHECK_INTERFACE(IAudioFlinger, data, reply); uint32_t sampleRate = data.readInt32(); - int format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); int channelCount = data.readInt32(); reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) ); return NO_ERROR; @@ -834,7 +834,7 @@ status_t BnAudioFlinger::onTransact( CHECK_INTERFACE(IAudioFlinger, data, reply); uint32_t devices = data.readInt32(); uint32_t samplingRate = data.readInt32(); - uint32_t format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); uint32_t channels = data.readInt32(); uint32_t latency = data.readInt32(); uint32_t flags = data.readInt32(); @@ -879,7 +879,7 @@ status_t BnAudioFlinger::onTransact( CHECK_INTERFACE(IAudioFlinger, data, reply); uint32_t devices = data.readInt32(); uint32_t samplingRate = data.readInt32(); - uint32_t format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); uint32_t channels = data.readInt32(); uint32_t acoutics = data.readInt32(); diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index e363101..b5c857f 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -122,7 +122,7 @@ public: virtual audio_io_handle_t getOutput( audio_stream_type_t stream, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_policy_output_flags_t flags) { @@ -174,7 +174,7 @@ public: virtual audio_io_handle_t getInput( int inputSource, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_in_acoustics_t acoustics, int audioSession) @@ -416,7 +416,7 @@ status_t BnAudioPolicyService::onTransact( audio_stream_type_t stream = static_cast <audio_stream_type_t>(data.readInt32()); uint32_t samplingRate = data.readInt32(); - uint32_t format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); uint32_t channels = data.readInt32(); audio_policy_output_flags_t flags = static_cast <audio_policy_output_flags_t>(data.readInt32()); @@ -463,7 +463,7 @@ status_t BnAudioPolicyService::onTransact( CHECK_INTERFACE(IAudioPolicyService, data, reply); int inputSource = data.readInt32(); uint32_t samplingRate = data.readInt32(); - uint32_t format = data.readInt32(); + audio_format_t format = (audio_format_t) data.readInt32(); uint32_t channels = data.readInt32(); audio_in_acoustics_t acoustics = static_cast <audio_in_acoustics_t>(data.readInt32()); diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 53b13f7..c6a9c77 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -382,7 +382,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -502,13 +502,13 @@ int AudioFlinger::channelCount(int output) const return thread->channelCount(); } -uint32_t AudioFlinger::format(int output) const +audio_format_t AudioFlinger::format(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("format() unknown thread %d", output); - return 0; + return AUDIO_FORMAT_INVALID; } return thread->format(); } @@ -849,7 +849,7 @@ String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) return String8(""); } -size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) +size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) { status_t ret = initCheck(); if (ret != NO_ERROR) { @@ -990,7 +990,7 @@ void AudioFlinger::removeClient_l(pid_t pid) AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) : Thread(false), mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), - mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), + mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), mDevice(device) { mDeathRecipient = new PMDeathRecipient(this); @@ -1033,7 +1033,7 @@ int AudioFlinger::ThreadBase::channelCount() const return (int)mChannelCount; } -uint32_t AudioFlinger::ThreadBase::format() const +audio_format_t AudioFlinger::ThreadBase::format() const { return mFormat; } @@ -1495,7 +1495,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra const sp<AudioFlinger::Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, @@ -2394,7 +2394,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - if (value != AUDIO_FORMAT_PCM_16_BIT) { + if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { status = BAD_VALUE; } else { reconfig = true; @@ -3233,7 +3233,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -3395,7 +3395,7 @@ AudioFlinger::PlaybackThread::Track::Track( const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, @@ -3701,7 +3701,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -3814,7 +3814,7 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount) : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), @@ -4147,7 +4147,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( pid_t pid, int input, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -4492,7 +4492,7 @@ bool AudioFlinger::RecordThread::threadLoop() sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, - int format, + audio_format_t format, int channelMask, int frameCount, uint32_t flags, @@ -4704,7 +4704,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; - int reqFormat = mFormat; + audio_format_t reqFormat = mFormat; int reqSamplingRate = mReqSampleRate; int reqChannelCount = mReqChannelCount; @@ -4713,7 +4713,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - reqFormat = value; + reqFormat = (audio_format_t) value; reconfig = true; } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { @@ -4924,7 +4924,7 @@ audio_stream_t* AudioFlinger::RecordThread::stream() int AudioFlinger::openOutput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, uint32_t flags) @@ -4933,7 +4933,7 @@ int AudioFlinger::openOutput(uint32_t *pDevices, PlaybackThread *thread = NULL; mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; + audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t channels = pChannels ? *pChannels : 0; uint32_t latency = pLatencyMs ? *pLatencyMs : 0; audio_stream_out_t *outStream; @@ -4956,7 +4956,7 @@ int AudioFlinger::openOutput(uint32_t *pDevices, if (outHwDev == NULL) return 0; - status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, + status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, &channels, &samplingRate, &outStream); ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", outStream, @@ -5084,17 +5084,17 @@ status_t AudioFlinger::restoreOutput(int output) int AudioFlinger::openInput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t acoustics) { status_t status; RecordThread *thread = NULL; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; + audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t channels = pChannels ? *pChannels : 0; uint32_t reqSamplingRate = samplingRate; - uint32_t reqFormat = format; + audio_format_t reqFormat = format; uint32_t reqChannels = channels; audio_stream_in_t *inStream; audio_hw_device_t *inHwDev; @@ -5109,7 +5109,7 @@ int AudioFlinger::openInput(uint32_t *pDevices, if (inHwDev == NULL) return 0; - status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, + status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, &channels, &samplingRate, (audio_in_acoustics_t)acoustics, &inStream); @@ -5129,7 +5129,7 @@ int AudioFlinger::openInput(uint32_t *pDevices, (samplingRate <= 2 * reqSamplingRate) && (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { ALOGV("openInput() reopening with proposed sampling rate and channels"); - status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, + status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, &channels, &samplingRate, (audio_in_acoustics_t)acoustics, &inStream); diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 8a82bdb..d862c1d 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -72,7 +72,7 @@ public: pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -83,7 +83,7 @@ public: virtual uint32_t sampleRate(int output) const; virtual int channelCount(int output) const; - virtual uint32_t format(int output) const; + virtual audio_format_t format(int output) const; virtual size_t frameCount(int output) const; virtual uint32_t latency(int output) const; @@ -109,12 +109,12 @@ public: virtual void registerClient(const sp<IAudioFlingerClient>& client); - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); + virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount); virtual unsigned int getInputFramesLost(int ioHandle); virtual int openOutput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, uint32_t flags); @@ -129,7 +129,7 @@ public: virtual int openInput(uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t acoustics); @@ -189,7 +189,7 @@ public: pid_t pid, int input, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -315,7 +315,7 @@ private: TrackBase(const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -343,7 +343,7 @@ private: virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0; virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - uint32_t format() const { + audio_format_t format() const { return mFormat; } @@ -376,7 +376,7 @@ private: // we don't really need a lock for these int mState; int mClientTid; - uint32_t mFormat; + audio_format_t mFormat; uint32_t mFlags; int mSessionId; uint8_t mChannelCount; @@ -410,7 +410,7 @@ private: int type() const { return mType; } uint32_t sampleRate() const; int channelCount() const; - uint32_t format() const; + audio_format_t format() const; size_t frameCount() const; void wakeUp() { mWaitWorkCV.broadcast(); } void exit(); @@ -537,7 +537,7 @@ private: uint32_t mChannelMask; uint16_t mChannelCount; size_t mFrameSize; - uint32_t mFormat; + audio_format_t mFormat; Condition mParamCond; Vector<String8> mNewParameters; status_t mParamStatus; @@ -575,7 +575,7 @@ private: const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, @@ -660,7 +660,7 @@ private: OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount); ~OutputTrack(); @@ -715,7 +715,7 @@ private: const sp<AudioFlinger::Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, const sp<IMemory>& sharedBuffer, @@ -935,7 +935,7 @@ private: RecordTrack(const wp<ThreadBase>& thread, const sp<Client>& client, uint32_t sampleRate, - uint32_t format, + audio_format_t format, uint32_t channelMask, int frameCount, uint32_t flags, @@ -979,7 +979,7 @@ private: sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, - int format, + audio_format_t format, int channelMask, int frameCount, uint32_t flags, diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 7a408bc..28b1c89 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -241,7 +241,7 @@ audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_policy_output_flags_t flags) { @@ -289,7 +289,7 @@ void AudioPolicyService::releaseOutput(audio_io_handle_t output) audio_io_handle_t AudioPolicyService::getInput(int inputSource, uint32_t samplingRate, - uint32_t format, + audio_format_t format, uint32_t channels, audio_in_acoustics_t acoustics, int audioSession) @@ -1352,7 +1352,7 @@ extern "C" { static audio_io_handle_t aps_open_output(void *service, uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, audio_policy_output_flags_t flags) @@ -1413,7 +1413,7 @@ static int aps_restore_output(void *service, audio_io_handle_t output) static audio_io_handle_t aps_open_input(void *service, uint32_t *pDevices, uint32_t *pSamplingRate, - uint32_t *pFormat, + audio_format_t *pFormat, uint32_t *pChannels, uint32_t acoustics) { diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h index 3693cef..9811670 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audioflinger/AudioPolicyService.h @@ -63,7 +63,7 @@ public: virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); virtual audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = 0, audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT); @@ -76,7 +76,7 @@ public: virtual void releaseOutput(audio_io_handle_t output); virtual audio_io_handle_t getInput(int inputSource, uint32_t samplingRate = 0, - uint32_t format = AUDIO_FORMAT_DEFAULT, + audio_format_t format = AUDIO_FORMAT_DEFAULT, uint32_t channels = 0, audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0, |