diff options
Diffstat (limited to 'media')
-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 |
4 files changed, 30 insertions, 30 deletions
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()); |