From fce7a473248381cc83a01855f92581077d3c9ee2 Mon Sep 17 00:00:00 2001 From: Dima Zavin Date: Tue, 19 Apr 2011 22:30:36 -0700 Subject: audio/media: convert to using the audio HAL and new audio defs Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5 Signed-off-by: Dima Zavin --- media/libmedia/AudioRecord.cpp | 23 ++- media/libmedia/AudioSystem.cpp | 207 +++++---------------- media/libmedia/AudioTrack.cpp | 49 ++--- media/libmedia/IAudioPolicyService.cpp | 94 +++++----- media/libmedia/JetPlayer.cpp | 4 +- media/libmedia/ToneGenerator.cpp | 4 +- media/libmedia/Visualizer.cpp | 4 +- media/libmedia/mediaplayer.cpp | 4 +- media/libmediaplayerservice/MediaPlayerService.cpp | 8 +- media/libmediaplayerservice/MediaPlayerService.h | 4 +- .../libmediaplayerservice/MediaRecorderClient.cpp | 4 +- media/libmediaplayerservice/MidiFile.cpp | 6 +- .../libmediaplayerservice/StagefrightRecorder.cpp | 30 +-- media/libmediaplayerservice/StagefrightRecorder.h | 6 +- media/libstagefright/AudioPlayer.cpp | 8 +- media/libstagefright/AudioSource.cpp | 4 +- 16 files changed, 186 insertions(+), 273 deletions(-) (limited to 'media') diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 5d74a0a..8438714 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -37,6 +37,9 @@ #include #include +#include +#include + #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) @@ -66,8 +69,8 @@ status_t AudioRecord::getMinFrameCount( // We double the size of input buffer for ping pong use of record buffer. size <<= 1; - if (AudioSystem::isLinearPCM(format)) { - size /= channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1); + if (audio_is_linear_pcm(format)) { + size /= channelCount * (format == AUDIO_FORMAT_PCM_16_BIT ? 2 : 1); } *frameCount = size; @@ -145,22 +148,22 @@ status_t AudioRecord::set( } // these below should probably come from the audioFlinger too... if (format == 0) { - format = AudioSystem::PCM_16_BIT; + format = AUDIO_FORMAT_PCM_16_BIT; } // validate parameters - if (!AudioSystem::isValidFormat(format)) { + if (!audio_is_valid_format(format)) { LOGE("Invalid format"); return BAD_VALUE; } - if (!AudioSystem::isInputChannel(channels)) { + if (!audio_is_input_channel(channels)) { return BAD_VALUE; } - int channelCount = AudioSystem::popCount(channels); + int channelCount = popcount(channels); audio_io_handle_t input = AudioSystem::getInput(inputSource, - sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags); + sampleRate, format, channels, (audio_in_acoustics_t)flags); if (input == 0) { LOGE("Could not get audio input for record source %d", inputSource); return BAD_VALUE; @@ -254,8 +257,8 @@ uint32_t AudioRecord::frameCount() const int AudioRecord::frameSize() const { - if (AudioSystem::isLinearPCM(mFormat)) { - return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); + if (audio_is_linear_pcm(mFormat)) { + return channelCount()*((format() == AUDIO_FORMAT_PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); } else { return sizeof(uint8_t); } @@ -587,7 +590,7 @@ audio_io_handle_t AudioRecord::getInput_l() mInput = AudioSystem::getInput(mInputSource, mCblk->sampleRate, mFormat, mChannels, - (AudioSystem::audio_in_acoustics)mFlags); + (audio_in_acoustics_t)mFlags); return mInput; } diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 5c6f344..e08a55b 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -23,6 +23,8 @@ #include #include +#include + // ---------------------------------------------------------------------------- // the sim build doesn't have gettid @@ -45,7 +47,7 @@ DefaultKeyedVector AudioSyst // Cached values for recording queries uint32_t AudioSystem::gPrevInSamplingRate = 16000; -int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT; +int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT; int AudioSystem::gPrevInChannelCount = 1; size_t AudioSystem::gInBuffSize = 0; @@ -127,7 +129,7 @@ status_t AudioSystem::getMasterMute(bool* mute) status_t AudioSystem::setStreamVolume(int stream, float value, int output) { - if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE; + if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE; const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; af->setStreamVolume(stream, value, output); @@ -136,7 +138,7 @@ status_t AudioSystem::setStreamVolume(int stream, float value, int output) status_t AudioSystem::setStreamMute(int stream, bool mute) { - if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE; + if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE; const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; af->setStreamMute(stream, mute); @@ -145,7 +147,7 @@ status_t AudioSystem::setStreamMute(int stream, bool mute) status_t AudioSystem::getStreamVolume(int stream, float* volume, int output) { - if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE; + if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE; const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; *volume = af->streamVolume(stream, output); @@ -154,7 +156,7 @@ status_t AudioSystem::getStreamVolume(int stream, float* volume, int output) status_t AudioSystem::getStreamMute(int stream, bool* mute) { - if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE; + if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE; const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; *mute = af->streamMute(stream); @@ -163,7 +165,7 @@ status_t AudioSystem::getStreamMute(int stream, bool* mute) status_t AudioSystem::setMode(int mode) { - if (mode >= NUM_MODES) return BAD_VALUE; + if (mode >= AUDIO_MODE_CNT) return BAD_VALUE; const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; return af->setMode(mode); @@ -213,11 +215,11 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) OutputDescriptor *outputDesc; audio_io_handle_t output; - if (streamType == DEFAULT) { - streamType = MUSIC; + if (streamType == AUDIO_STREAM_DEFAULT) { + streamType = AUDIO_STREAM_MUSIC; } - output = getOutput((stream_type)streamType); + output = getOutput((audio_stream_type_t)streamType); if (output == 0) { return PERMISSION_DENIED; } @@ -246,11 +248,11 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) OutputDescriptor *outputDesc; audio_io_handle_t output; - if (streamType == DEFAULT) { - streamType = MUSIC; + if (streamType == AUDIO_STREAM_DEFAULT) { + streamType = AUDIO_STREAM_MUSIC; } - output = getOutput((stream_type)streamType); + output = getOutput((audio_stream_type_t)streamType); if (output == 0) { return PERMISSION_DENIED; } @@ -277,11 +279,11 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType) OutputDescriptor *outputDesc; audio_io_handle_t output; - if (streamType == DEFAULT) { - streamType = MUSIC; + if (streamType == AUDIO_STREAM_DEFAULT) { + streamType = AUDIO_STREAM_MUSIC; } - output = getOutput((stream_type)streamType); + output = getOutput((audio_stream_type_t)streamType); if (output == 0) { return PERMISSION_DENIED; } @@ -338,11 +340,11 @@ status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames const sp& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; - if (stream == DEFAULT) { - stream = MUSIC; + if (stream == AUDIO_STREAM_DEFAULT) { + stream = AUDIO_STREAM_MUSIC; } - return af->getRenderPosition(halFrames, dspFrames, getOutput((stream_type)stream)); + return af->getRenderPosition(halFrames, dspFrames, getOutput((audio_stream_type_t)stream)); } unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { @@ -455,10 +457,10 @@ void AudioSystem::setErrorCallback(audio_error_callback cb) { bool AudioSystem::routedToA2dpOutput(int streamType) { switch(streamType) { - case MUSIC: - case VOICE_CALL: - case BLUETOOTH_SCO: - case SYSTEM: + case AUDIO_STREAM_MUSIC: + case AUDIO_STREAM_VOICE_CALL: + case AUDIO_STREAM_BLUETOOTH_SCO: + case AUDIO_STREAM_SYSTEM: return true; default: return false; @@ -497,9 +499,9 @@ const sp& AudioSystem::get_audio_policy_service() return gAudioPolicyService; } -status_t AudioSystem::setDeviceConnectionState(audio_devices device, - device_connection_state state, - const char *device_address) +status_t AudioSystem::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return PERMISSION_DENIED; @@ -507,11 +509,11 @@ status_t AudioSystem::setDeviceConnectionState(audio_devices device, return aps->setDeviceConnectionState(device, state, device_address); } -AudioSystem::device_connection_state AudioSystem::getDeviceConnectionState(audio_devices device, +audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device, const char *device_address) { const sp& aps = AudioSystem::get_audio_policy_service(); - if (aps == 0) return DEVICE_STATE_UNAVAILABLE; + if (aps == 0) return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; return aps->getDeviceConnectionState(device, device_address); } @@ -531,26 +533,26 @@ status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask) return aps->setRingerMode(mode, mask); } -status_t AudioSystem::setForceUse(force_use usage, forced_config config) +status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return PERMISSION_DENIED; return aps->setForceUse(usage, config); } -AudioSystem::forced_config AudioSystem::getForceUse(force_use usage) +audio_policy_forced_cfg_t AudioSystem::getForceUse(audio_policy_force_use_t usage) { const sp& aps = AudioSystem::get_audio_policy_service(); - if (aps == 0) return FORCE_NONE; + if (aps == 0) return AUDIO_POLICY_FORCE_NONE; return aps->getForceUse(usage); } -audio_io_handle_t AudioSystem::getOutput(stream_type stream, +audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream, uint32_t samplingRate, uint32_t format, uint32_t channels, - output_flags flags) + audio_policy_output_flags_t flags) { audio_io_handle_t output = 0; // Do not use stream to output map cache if the direct output @@ -561,9 +563,9 @@ audio_io_handle_t AudioSystem::getOutput(stream_type stream, // be reworked for proper operation with direct outputs. This code is too specific // to the first use case we want to cover (Voice Recognition and Voice Dialer over // Bluetooth SCO - if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0 && - ((stream != AudioSystem::VOICE_CALL && stream != AudioSystem::BLUETOOTH_SCO) || - channels != AudioSystem::CHANNEL_OUT_MONO || + if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) == 0 && + ((stream != AUDIO_STREAM_VOICE_CALL && stream != AUDIO_STREAM_BLUETOOTH_SCO) || + channels != AUDIO_CHANNEL_OUT_MONO || (samplingRate != 8000 && samplingRate != 16000))) { Mutex::Autolock _l(gLock); output = AudioSystem::gStreamOutputMap.valueFor(stream); @@ -573,7 +575,7 @@ audio_io_handle_t AudioSystem::getOutput(stream_type stream, const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return 0; output = aps->getOutput(stream, samplingRate, format, channels, flags); - if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) { + if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) == 0) { Mutex::Autolock _l(gLock); AudioSystem::gStreamOutputMap.add(stream, output); } @@ -582,7 +584,7 @@ audio_io_handle_t AudioSystem::getOutput(stream_type stream, } status_t AudioSystem::startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, + audio_stream_type_t stream, int session) { const sp& aps = AudioSystem::get_audio_policy_service(); @@ -591,7 +593,7 @@ status_t AudioSystem::startOutput(audio_io_handle_t output, } status_t AudioSystem::stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, + audio_stream_type_t stream, int session) { const sp& aps = AudioSystem::get_audio_policy_service(); @@ -610,7 +612,7 @@ audio_io_handle_t AudioSystem::getInput(int inputSource, uint32_t samplingRate, uint32_t format, uint32_t channels, - audio_in_acoustics acoustics) + audio_in_acoustics_t acoustics) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return 0; @@ -638,7 +640,7 @@ void AudioSystem::releaseInput(audio_io_handle_t input) aps->releaseInput(input); } -status_t AudioSystem::initStreamVolume(stream_type stream, +status_t AudioSystem::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) { @@ -647,28 +649,28 @@ status_t AudioSystem::initStreamVolume(stream_type stream, return aps->initStreamVolume(stream, indexMin, indexMax); } -status_t AudioSystem::setStreamVolumeIndex(stream_type stream, int index) +status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream, int index) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return PERMISSION_DENIED; return aps->setStreamVolumeIndex(stream, index); } -status_t AudioSystem::getStreamVolumeIndex(stream_type stream, int *index) +status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream, int *index) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return PERMISSION_DENIED; return aps->getStreamVolumeIndex(stream, index); } -uint32_t AudioSystem::getStrategyForStream(AudioSystem::stream_type stream) +uint32_t AudioSystem::getStrategyForStream(audio_stream_type_t stream) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return 0; return aps->getStrategyForStream(stream); } -uint32_t AudioSystem::getDevicesForStream(AudioSystem::stream_type stream) +uint32_t AudioSystem::getDevicesForStream(audio_stream_type_t stream) { const sp& aps = AudioSystem::get_audio_policy_service(); if (aps == 0) return 0; @@ -717,122 +719,5 @@ void AudioSystem::AudioPolicyServiceClient::binderDied(const wp& who) { LOGW("AudioPolicyService server died!"); } -// --------------------------------------------------------------------------- - - -// use emulated popcount optimization -// http://www.df.lth.se/~john_e/gems/gem002d.html -uint32_t AudioSystem::popCount(uint32_t u) -{ - u = ((u&0x55555555) + ((u>>1)&0x55555555)); - u = ((u&0x33333333) + ((u>>2)&0x33333333)); - u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f)); - u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff)); - u = ( u&0x0000ffff) + (u>>16); - return u; -} - -bool AudioSystem::isOutputDevice(audio_devices device) -{ - if ((popCount(device) == 1 ) && - ((device & ~AudioSystem::DEVICE_OUT_ALL) == 0)) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isInputDevice(audio_devices device) -{ - if ((popCount(device) == 1 ) && - ((device & ~AudioSystem::DEVICE_IN_ALL) == 0)) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isA2dpDevice(audio_devices device) -{ - if ((popCount(device) == 1 ) && - (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP | - AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | - AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER))) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isBluetoothScoDevice(audio_devices device) -{ - if ((popCount(device) == 1 ) && - (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO | - AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET | - AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT | - AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET))) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isLowVisibility(stream_type stream) -{ - if (stream == AudioSystem::SYSTEM || - stream == AudioSystem::NOTIFICATION || - stream == AudioSystem::RING) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isInputChannel(uint32_t channel) -{ - if ((channel & ~AudioSystem::CHANNEL_IN_ALL) == 0) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isOutputChannel(uint32_t channel) -{ - if ((channel & ~AudioSystem::CHANNEL_OUT_ALL) == 0) { - return true; - } else { - return false; - } -} - -bool AudioSystem::isValidFormat(uint32_t format) -{ - switch (format & MAIN_FORMAT_MASK) { - case PCM: - case MP3: - case AMR_NB: - case AMR_WB: - case AAC: - case HE_AAC_V1: - case HE_AAC_V2: - case VORBIS: - return true; - default: - return false; - } -} - -bool AudioSystem::isLinearPCM(uint32_t format) -{ - switch (format) { - case PCM_16_BIT: - case PCM_8_BIT: - return true; - default: - return false; - } -} - }; // namespace android diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 66e11d2..2673df9 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -37,6 +37,11 @@ #include #include +#include + +#include +#include + #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) @@ -165,39 +170,41 @@ status_t AudioTrack::set( } // handle default values first. - if (streamType == AudioSystem::DEFAULT) { - streamType = AudioSystem::MUSIC; + if (streamType == AUDIO_STREAM_DEFAULT) { + streamType = AUDIO_STREAM_MUSIC; } if (sampleRate == 0) { sampleRate = afSampleRate; } // these below should probably come from the audioFlinger too... if (format == 0) { - format = AudioSystem::PCM_16_BIT; + format = AUDIO_FORMAT_PCM_16_BIT; } if (channels == 0) { - channels = AudioSystem::CHANNEL_OUT_STEREO; + channels = AUDIO_CHANNEL_OUT_STEREO; } // validate parameters - if (!AudioSystem::isValidFormat(format)) { + if (!audio_is_valid_format(format)) { LOGE("Invalid format"); return BAD_VALUE; } // force direct flag if format is not linear PCM - if (!AudioSystem::isLinearPCM(format)) { - flags |= AudioSystem::OUTPUT_FLAG_DIRECT; + if (!audio_is_linear_pcm(format)) { + flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT; } - if (!AudioSystem::isOutputChannel(channels)) { + if (!audio_is_output_channel(channels)) { LOGE("Invalid channel mask"); return BAD_VALUE; } - uint32_t channelCount = AudioSystem::popCount(channels); + uint32_t channelCount = popcount(channels); - audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType, - sampleRate, format, channels, (AudioSystem::output_flags)flags); + audio_io_handle_t output = AudioSystem::getOutput( + (audio_stream_type_t)streamType, + sampleRate,format, channels, + (audio_policy_output_flags_t)flags); if (output == 0) { LOGE("Could not get audio output for stream type %d", streamType); @@ -290,8 +297,8 @@ uint32_t AudioTrack::frameCount() const int AudioTrack::frameSize() const { - if (AudioSystem::isLinearPCM(mFormat)) { - return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); + if (audio_is_linear_pcm(mFormat)) { + return channelCount()*((format() == AUDIO_FORMAT_PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); } else { return sizeof(uint8_t); } @@ -673,8 +680,8 @@ audio_io_handle_t AudioTrack::getOutput() // must be called with mLock held audio_io_handle_t AudioTrack::getOutput_l() { - return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType, - mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags); + return AudioSystem::getOutput((audio_stream_type_t)mStreamType, + mCblk->sampleRate, mFormat, mChannels, (audio_policy_output_flags_t)mFlags); } int AudioTrack::getSessionId() @@ -727,7 +734,7 @@ status_t AudioTrack::createTrack_l( } mNotificationFramesAct = mNotificationFramesReq; - if (!AudioSystem::isLinearPCM(format)) { + if (!audio_is_linear_pcm(format)) { if (sharedBuffer != 0) { frameCount = sharedBuffer->size(); } @@ -923,8 +930,8 @@ create_new_track: audioBuffer->channelCount = mChannelCount; audioBuffer->frameCount = framesReq; audioBuffer->size = framesReq * cblk->frameSize; - if (AudioSystem::isLinearPCM(mFormat)) { - audioBuffer->format = AudioSystem::PCM_16_BIT; + if (audio_is_linear_pcm(mFormat)) { + audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; } else { audioBuffer->format = mFormat; } @@ -982,7 +989,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) size_t toWrite; - if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { + if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { // Divide capacity by 2 to take expansion into account toWrite = audioBuffer.size>>1; // 8 to 16 bit conversion @@ -1085,7 +1092,7 @@ bool AudioTrack::processAudioBuffer(const sp& thread) // Divide buffer size by 2 to take into account the expansion // due to 8 to 16 bit conversion: the callback must fill only half // of the destination buffer - if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { + if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { audioBuffer.size >>= 1; } @@ -1104,7 +1111,7 @@ bool AudioTrack::processAudioBuffer(const sp& thread) } if (writtenSize > reqSize) writtenSize = reqSize; - if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { + if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) { // 8 to 16 bit conversion const int8_t *src = audioBuffer.i8 + writtenSize-1; int count = writtenSize; diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index b89a278..88a9ae0 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -25,6 +25,8 @@ #include +#include + namespace android { enum { @@ -62,8 +64,8 @@ public: } virtual status_t setDeviceConnectionState( - AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, + audio_devices_t device, + audio_policy_dev_state_t state, const char *device_address) { Parcel data, reply; @@ -75,8 +77,8 @@ public: return static_cast (reply.readInt32()); } - virtual AudioSystem::device_connection_state getDeviceConnectionState( - AudioSystem::audio_devices device, + virtual audio_policy_dev_state_t getDeviceConnectionState( + audio_devices_t device, const char *device_address) { Parcel data, reply; @@ -84,7 +86,7 @@ public: data.writeInt32(static_cast (device)); data.writeCString(device_address); remote()->transact(GET_DEVICE_CONNECTION_STATE, data, &reply); - return static_cast (reply.readInt32()); + return static_cast (reply.readInt32()); } virtual status_t setPhoneState(int state) @@ -106,7 +108,7 @@ public: return static_cast (reply.readInt32()); } - virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) + virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -116,21 +118,21 @@ public: return static_cast (reply.readInt32()); } - virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) + virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); data.writeInt32(static_cast (usage)); remote()->transact(GET_FORCE_USE, data, &reply); - return static_cast (reply.readInt32()); + return static_cast (reply.readInt32()); } virtual audio_io_handle_t getOutput( - AudioSystem::stream_type stream, + audio_stream_type_t stream, uint32_t samplingRate, uint32_t format, uint32_t channels, - AudioSystem::output_flags flags) + audio_policy_output_flags_t flags) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -144,7 +146,7 @@ public: } virtual status_t startOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, + audio_stream_type_t stream, int session) { Parcel data, reply; @@ -157,7 +159,7 @@ public: } virtual status_t stopOutput(audio_io_handle_t output, - AudioSystem::stream_type stream, + audio_stream_type_t stream, int session) { Parcel data, reply; @@ -182,7 +184,7 @@ public: uint32_t samplingRate, uint32_t format, uint32_t channels, - AudioSystem::audio_in_acoustics acoustics) + audio_in_acoustics_t acoustics) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -221,7 +223,7 @@ public: remote()->transact(RELEASE_INPUT, data, &reply); } - virtual status_t initStreamVolume(AudioSystem::stream_type stream, + virtual status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) { @@ -234,7 +236,7 @@ public: return static_cast (reply.readInt32()); } - virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) + virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -244,7 +246,7 @@ public: return static_cast (reply.readInt32()); } - virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) + virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -255,7 +257,7 @@ public: return static_cast (reply.readInt32()); } - virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) + virtual uint32_t getStrategyForStream(audio_stream_type_t stream) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -264,7 +266,7 @@ public: return reply.readInt32(); } - virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream) + virtual uint32_t getDevicesForStream(audio_stream_type_t stream) { Parcel data, reply; data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor()); @@ -330,10 +332,10 @@ status_t BnAudioPolicyService::onTransact( switch(code) { case SET_DEVICE_CONNECTION_STATE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::audio_devices device = - static_cast (data.readInt32()); - AudioSystem::device_connection_state state = - static_cast (data.readInt32()); + audio_devices_t device = + static_cast (data.readInt32()); + audio_policy_dev_state_t state = + static_cast (data.readInt32()); const char *device_address = data.readCString(); reply->writeInt32(static_cast (setDeviceConnectionState(device, state, @@ -343,8 +345,8 @@ status_t BnAudioPolicyService::onTransact( case GET_DEVICE_CONNECTION_STATE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::audio_devices device = - static_cast (data.readInt32()); + audio_devices_t device = + static_cast (data.readInt32()); const char *device_address = data.readCString(); reply->writeInt32(static_cast (getDeviceConnectionState(device, device_address))); @@ -367,29 +369,29 @@ status_t BnAudioPolicyService::onTransact( case SET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::force_use usage = static_cast (data.readInt32()); - AudioSystem::forced_config config = - static_cast (data.readInt32()); + audio_policy_force_use_t usage = static_cast (data.readInt32()); + audio_policy_forced_cfg_t config = + static_cast (data.readInt32()); reply->writeInt32(static_cast (setForceUse(usage, config))); return NO_ERROR; } break; case GET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::force_use usage = static_cast (data.readInt32()); + audio_policy_force_use_t usage = static_cast (data.readInt32()); reply->writeInt32(static_cast (getForceUse(usage))); return NO_ERROR; } break; case GET_OUTPUT: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); uint32_t samplingRate = data.readInt32(); uint32_t format = data.readInt32(); uint32_t channels = data.readInt32(); - AudioSystem::output_flags flags = - static_cast (data.readInt32()); + audio_policy_output_flags_t flags = + static_cast (data.readInt32()); audio_io_handle_t output = getOutput(stream, samplingRate, @@ -406,7 +408,7 @@ status_t BnAudioPolicyService::onTransact( uint32_t stream = data.readInt32(); int session = data.readInt32(); reply->writeInt32(static_cast (startOutput(output, - (AudioSystem::stream_type)stream, + (audio_stream_type_t)stream, session))); return NO_ERROR; } break; @@ -417,7 +419,7 @@ status_t BnAudioPolicyService::onTransact( uint32_t stream = data.readInt32(); int session = data.readInt32(); reply->writeInt32(static_cast (stopOutput(output, - (AudioSystem::stream_type)stream, + (audio_stream_type_t)stream, session))); return NO_ERROR; } break; @@ -435,8 +437,8 @@ status_t BnAudioPolicyService::onTransact( uint32_t samplingRate = data.readInt32(); uint32_t format = data.readInt32(); uint32_t channels = data.readInt32(); - AudioSystem::audio_in_acoustics acoustics = - static_cast (data.readInt32()); + audio_in_acoustics_t acoustics = + static_cast (data.readInt32()); audio_io_handle_t input = getInput(inputSource, samplingRate, format, @@ -469,8 +471,8 @@ status_t BnAudioPolicyService::onTransact( case INIT_STREAM_VOLUME: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); int indexMin = data.readInt32(); int indexMax = data.readInt32(); reply->writeInt32(static_cast (initStreamVolume(stream, indexMin,indexMax))); @@ -479,8 +481,8 @@ status_t BnAudioPolicyService::onTransact( case SET_STREAM_VOLUME: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); int index = data.readInt32(); reply->writeInt32(static_cast (setStreamVolumeIndex(stream, index))); return NO_ERROR; @@ -488,8 +490,8 @@ status_t BnAudioPolicyService::onTransact( case GET_STREAM_VOLUME: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); int index; status_t status = getStreamVolumeIndex(stream, &index); reply->writeInt32(index); @@ -499,16 +501,16 @@ status_t BnAudioPolicyService::onTransact( case GET_STRATEGY_FOR_STREAM: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); reply->writeInt32(getStrategyForStream(stream)); return NO_ERROR; } break; case GET_DEVICES_FOR_STREAM: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - AudioSystem::stream_type stream = - static_cast (data.readInt32()); + audio_stream_type_t stream = + static_cast (data.readInt32()); reply->writeInt32(static_cast (getDevicesForStream(stream))); return NO_ERROR; } break; diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp index ee9e1d8..88157d2 100644 --- a/media/libmedia/JetPlayer.cpp +++ b/media/libmedia/JetPlayer.cpp @@ -96,10 +96,10 @@ int JetPlayer::init() // create the output AudioTrack mAudioTrack = new AudioTrack(); - mAudioTrack->set(AudioSystem::MUSIC, //TODO parametrize this + mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parametrize this pLibConfig->sampleRate, 1, // format = PCM 16bits per sample, - (pLibConfig->numChannels == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO, + (pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO, mTrackBufferSize, 0); diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp index 82fe2d4..9f1b3d6 100644 --- a/media/libmedia/ToneGenerator.cpp +++ b/media/libmedia/ToneGenerator.cpp @@ -1026,8 +1026,8 @@ bool ToneGenerator::initAudioTrack() { mpAudioTrack->set(mStreamType, 0, - AudioSystem::PCM_16_BIT, - AudioSystem::CHANNEL_OUT_MONO, + AUDIO_FORMAT_PCM_16_BIT, + AUDIO_CHANNEL_OUT_MONO, 0, 0, audioCallback, diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp index 43571cf..366707c 100644 --- a/media/libmedia/Visualizer.cpp +++ b/media/libmedia/Visualizer.cpp @@ -24,6 +24,8 @@ #include #include +#include + #include extern void fixed_fft_real(int n, int32_t *v); @@ -127,7 +129,7 @@ status_t Visualizer::setCaptureSize(uint32_t size) { if (size > VISUALIZER_CAPTURE_SIZE_MAX || size < VISUALIZER_CAPTURE_SIZE_MIN || - AudioSystem::popCount(size) != 1) { + popcount(size) != 1) { return BAD_VALUE; } diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp index e80e742..9daa80f 100644 --- a/media/libmedia/mediaplayer.cpp +++ b/media/libmedia/mediaplayer.cpp @@ -37,6 +37,8 @@ #include #include +#include + namespace android { MediaPlayer::MediaPlayer() @@ -45,7 +47,7 @@ MediaPlayer::MediaPlayer() mListener = NULL; mCookie = NULL; mDuration = -1; - mStreamType = AudioSystem::MUSIC; + mStreamType = AUDIO_STREAM_MUSIC; mCurrentPosition = -1; mSeekPosition = -1; mCurrentState = MEDIA_PLAYER_IDLE; diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp index 6b97708..9dd353b 100644 --- a/media/libmediaplayerservice/MediaPlayerService.cpp +++ b/media/libmediaplayerservice/MediaPlayerService.cpp @@ -53,6 +53,8 @@ #include #include +#include + #include #include "MediaRecorderClient.h" @@ -1209,7 +1211,7 @@ MediaPlayerService::AudioOutput::AudioOutput(int sessionId) mSessionId(sessionId) { LOGV("AudioOutput(%d)", sessionId); mTrack = 0; - mStreamType = AudioSystem::MUSIC; + mStreamType = AUDIO_STREAM_MUSIC; mLeftVolume = 1.0; mRightVolume = 1.0; mLatency = 0; @@ -1319,7 +1321,7 @@ status_t MediaPlayerService::AudioOutput::open( mStreamType, sampleRate, format, - (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO, + (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO, frameCount, 0 /* flags */, CallbackWrapper, @@ -1331,7 +1333,7 @@ status_t MediaPlayerService::AudioOutput::open( mStreamType, sampleRate, format, - (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO, + (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO, frameCount, 0, NULL, diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h index 5539a37..31b518e 100644 --- a/media/libmediaplayerservice/MediaPlayerService.h +++ b/media/libmediaplayerservice/MediaPlayerService.h @@ -30,6 +30,8 @@ #include #include +#include + namespace android { class IMediaRecorder; @@ -130,7 +132,7 @@ class MediaPlayerService : public BnMediaPlayerService virtual ssize_t bufferSize() const { return frameSize() * mFrameCount; } virtual ssize_t frameCount() const { return mFrameCount; } virtual ssize_t channelCount() const { return (ssize_t)mChannelCount; } - virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); } + virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); } virtual uint32_t latency() const; virtual float msecsPerFrame() const; virtual status_t getPosition(uint32_t *position); diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp index 1a1780c..5a47384 100644 --- a/media/libmediaplayerservice/MediaRecorderClient.cpp +++ b/media/libmediaplayerservice/MediaRecorderClient.cpp @@ -35,6 +35,8 @@ #include +#include + #include "MediaRecorderClient.h" #include "MediaPlayerService.h" @@ -102,7 +104,7 @@ status_t MediaRecorderClient::setAudioSource(int as) LOGE("recorder is not initialized"); return NO_INIT; } - return mRecorder->setAudioSource((audio_source)as); + return mRecorder->setAudioSource((audio_source_t)as); } status_t MediaRecorderClient::setOutputFormat(int of) diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp index 1b0b05f..37a3db3 100644 --- a/media/libmediaplayerservice/MidiFile.cpp +++ b/media/libmediaplayerservice/MidiFile.cpp @@ -30,6 +30,8 @@ #include #include +#include + #include "MidiFile.h" #ifdef HAVE_GETTID @@ -58,7 +60,7 @@ static const S_EAS_LIB_CONFIG* pLibConfig = NULL; MidiFile::MidiFile() : mEasData(NULL), mEasHandle(NULL), mAudioBuffer(NULL), mPlayTime(-1), mDuration(-1), mState(EAS_STATE_ERROR), - mStreamType(AudioSystem::MUSIC), mLoop(false), mExit(false), + mStreamType(AUDIO_STREAM_MUSIC), mLoop(false), mExit(false), mPaused(false), mRender(false), mTid(-1) { LOGV("constructor"); @@ -423,7 +425,7 @@ status_t MidiFile::setLooping(int loop) } status_t MidiFile::createOutputTrack() { - if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels, AudioSystem::PCM_16_BIT, 2) != NO_ERROR) { + if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels, AUDIO_FORMAT_PCM_16_BIT, 2) != NO_ERROR) { LOGE("mAudioSink open failed"); return ERROR_OPEN_FAILED; } diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp index e3dfabb..01fbea1 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.cpp +++ b/media/libmediaplayerservice/StagefrightRecorder.cpp @@ -46,6 +46,8 @@ #include #include +#include + #include "ARTPWriter.h" namespace android { @@ -64,7 +66,7 @@ static void addBatteryData(uint32_t params) { StagefrightRecorder::StagefrightRecorder() : mWriter(NULL), mWriterAux(NULL), mOutputFd(-1), mOutputFdAux(-1), - mAudioSource(AUDIO_SOURCE_LIST_END), + mAudioSource(AUDIO_SOURCE_CNT), mVideoSource(VIDEO_SOURCE_LIST_END), mStarted(false) { @@ -82,10 +84,10 @@ status_t StagefrightRecorder::init() { return OK; } -status_t StagefrightRecorder::setAudioSource(audio_source as) { +status_t StagefrightRecorder::setAudioSource(audio_source_t as) { LOGV("setAudioSource: %d", as); if (as < AUDIO_SOURCE_DEFAULT || - as >= AUDIO_SOURCE_LIST_END) { + as >= AUDIO_SOURCE_CNT) { LOGE("Invalid audio source: %d", as); return BAD_VALUE; } @@ -800,7 +802,7 @@ status_t StagefrightRecorder::start() { mStarted = true; uint32_t params = IMediaPlayerService::kBatteryDataCodecStarted; - if (mAudioSource != AUDIO_SOURCE_LIST_END) { + if (mAudioSource != AUDIO_SOURCE_CNT) { params |= IMediaPlayerService::kBatteryDataTrackAudio; } if (mVideoSource != VIDEO_SOURCE_LIST_END) { @@ -874,7 +876,7 @@ status_t StagefrightRecorder::startAACRecording() { mOutputFormat == OUTPUT_FORMAT_AAC_ADTS); CHECK(mAudioEncoder == AUDIO_ENCODER_AAC); - CHECK(mAudioSource != AUDIO_SOURCE_LIST_END); + CHECK(mAudioSource != AUDIO_SOURCE_CNT); CHECK(0 == "AACWriter is not implemented yet"); @@ -900,7 +902,7 @@ status_t StagefrightRecorder::startAMRRecording() { } } - if (mAudioSource >= AUDIO_SOURCE_LIST_END) { + if (mAudioSource >= AUDIO_SOURCE_CNT) { LOGE("Invalid audio source: %d", mAudioSource); return BAD_VALUE; } @@ -933,9 +935,9 @@ status_t StagefrightRecorder::startAMRRecording() { status_t StagefrightRecorder::startRTPRecording() { CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP); - if ((mAudioSource != AUDIO_SOURCE_LIST_END + if ((mAudioSource != AUDIO_SOURCE_CNT && mVideoSource != VIDEO_SOURCE_LIST_END) - || (mAudioSource == AUDIO_SOURCE_LIST_END + || (mAudioSource == AUDIO_SOURCE_CNT && mVideoSource == VIDEO_SOURCE_LIST_END)) { // Must have exactly one source. return BAD_VALUE; @@ -947,7 +949,7 @@ status_t StagefrightRecorder::startRTPRecording() { sp source; - if (mAudioSource != AUDIO_SOURCE_LIST_END) { + if (mAudioSource != AUDIO_SOURCE_CNT) { source = createAudioSource(); } else { @@ -975,7 +977,7 @@ status_t StagefrightRecorder::startMPEG2TSRecording() { sp writer = new MPEG2TSWriter(mOutputFd); - if (mAudioSource != AUDIO_SOURCE_LIST_END) { + if (mAudioSource != AUDIO_SOURCE_CNT) { if (mAudioEncoder != AUDIO_ENCODER_AAC) { return ERROR_UNSUPPORTED; } @@ -1383,7 +1385,7 @@ status_t StagefrightRecorder::setupMPEG4Recording( // Audio source is added at the end if it exists. // This help make sure that the "recoding" sound is suppressed for // camcorder applications in the recorded files. - if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_LIST_END)) { + if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_CNT)) { err = setupAudioEncoder(writer); if (err != OK) return err; *totalBitRate += mAudioBitRate; @@ -1504,7 +1506,7 @@ status_t StagefrightRecorder::pause() { mStarted = false; uint32_t params = 0; - if (mAudioSource != AUDIO_SOURCE_LIST_END) { + if (mAudioSource != AUDIO_SOURCE_CNT) { params |= IMediaPlayerService::kBatteryDataTrackAudio; } if (mVideoSource != VIDEO_SOURCE_LIST_END) { @@ -1555,7 +1557,7 @@ status_t StagefrightRecorder::stop() { mStarted = false; uint32_t params = 0; - if (mAudioSource != AUDIO_SOURCE_LIST_END) { + if (mAudioSource != AUDIO_SOURCE_CNT) { params |= IMediaPlayerService::kBatteryDataTrackAudio; } if (mVideoSource != VIDEO_SOURCE_LIST_END) { @@ -1581,7 +1583,7 @@ status_t StagefrightRecorder::reset() { stop(); // No audio or video source by default - mAudioSource = AUDIO_SOURCE_LIST_END; + mAudioSource = AUDIO_SOURCE_CNT; mVideoSource = VIDEO_SOURCE_LIST_END; // Default parameters diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h index 2c440c1..3d463ea 100644 --- a/media/libmediaplayerservice/StagefrightRecorder.h +++ b/media/libmediaplayerservice/StagefrightRecorder.h @@ -22,6 +22,8 @@ #include #include +#include + namespace android { class Camera; @@ -39,7 +41,7 @@ struct StagefrightRecorder : public MediaRecorderBase { virtual ~StagefrightRecorder(); virtual status_t init(); - virtual status_t setAudioSource(audio_source as); + virtual status_t setAudioSource(audio_source_t as); virtual status_t setVideoSource(video_source vs); virtual status_t setOutputFormat(output_format of); virtual status_t setAudioEncoder(audio_encoder ae); @@ -69,7 +71,7 @@ private: sp mWriter, mWriterAux; sp mAudioSourceNode; - audio_source mAudioSource; + audio_source_t mAudioSource; video_source mVideoSource; output_format mOutputFormat; audio_encoder mAudioEncoder; diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp index fcea848..69f9c23 100644 --- a/media/libstagefright/AudioPlayer.cpp +++ b/media/libstagefright/AudioPlayer.cpp @@ -110,7 +110,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { if (mAudioSink.get() != NULL) { status_t err = mAudioSink->open( - mSampleRate, numChannels, AudioSystem::PCM_16_BIT, + mSampleRate, numChannels, AUDIO_FORMAT_PCM_16_BIT, DEFAULT_AUDIOSINK_BUFFERCOUNT, &AudioPlayer::AudioSinkCallback, this); if (err != OK) { @@ -132,10 +132,10 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { mAudioSink->start(); } else { mAudioTrack = new AudioTrack( - AudioSystem::MUSIC, mSampleRate, AudioSystem::PCM_16_BIT, + AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, (numChannels == 2) - ? AudioSystem::CHANNEL_OUT_STEREO - : AudioSystem::CHANNEL_OUT_MONO, + ? AUDIO_CHANNEL_OUT_STEREO + : AUDIO_CHANNEL_OUT_MONO, 0, 0, &AudioCallback, this, 0); if ((err = mAudioTrack->initCheck()) != OK) { diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp index bbdec02..99c3682 100644 --- a/media/libstagefright/AudioSource.cpp +++ b/media/libstagefright/AudioSource.cpp @@ -60,8 +60,8 @@ AudioSource::AudioSource( AudioRecord::RECORD_NS_ENABLE | AudioRecord::RECORD_IIR_ENABLE; mRecord = new AudioRecord( - inputSource, sampleRate, AudioSystem::PCM_16_BIT, - channels > 1? AudioSystem::CHANNEL_IN_STEREO: AudioSystem::CHANNEL_IN_MONO, + inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, + channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO, 4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */ flags, AudioRecordCallbackFunction, -- cgit v1.1