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Commit 254af180 authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Use audio_channel_mask_t more places 

Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.

Not done: AudioTrack and OutputDescriptor.

Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
parent 4d2132c8

include/media/AudioSystem.h

+2 −2
Original line number Diff line number Diff line
@@ -188,7 +188,7 @@ public: 
    static audio_io_handle_t getOutput(audio_stream_type_t stream,

                                        uint32_t samplingRate = 0,

                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,

                                        uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,

                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,

                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);

    static status_t startOutput(audio_io_handle_t output,

                                audio_stream_type_t stream,
@@ -200,7 +200,7 @@ public: 
    static audio_io_handle_t getInput(audio_source_t inputSource,

                                    uint32_t samplingRate = 0,

                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,

                                    uint32_t channels = AUDIO_CHANNEL_IN_MONO,

                                    audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,

                                    int sessionId = 0);

    static status_t startInput(audio_io_handle_t input);

    static status_t stopInput(audio_io_handle_t input);

include/media/IAudioPolicyService.h

+2 −2
Original line number Diff line number Diff line
@@ -51,7 +51,7 @@ public: 
    virtual audio_io_handle_t getOutput(audio_stream_type_t stream,

                                        uint32_t samplingRate = 0,

                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,

                                        uint32_t channels = 0,

                                        audio_channel_mask_t channelMask = 0,

                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;

    virtual status_t startOutput(audio_io_handle_t output,

                                 audio_stream_type_t stream,
@@ -63,7 +63,7 @@ public: 
    virtual audio_io_handle_t getInput(audio_source_t inputSource,

                                    uint32_t samplingRate = 0,

                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,

                                    uint32_t channels = 0,

                                    audio_channel_mask_t channelMask = 0,

                                    int audioSession = 0) = 0;

    virtual status_t startInput(audio_io_handle_t input) = 0;

    virtual status_t stopInput(audio_io_handle_t input) = 0;

media/libmedia/AudioSystem.cpp

+6 −6
Original line number Diff line number Diff line
@@ -449,7 +449,7 @@ 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 %d frameCount %d latency %d",

        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: {
@@ -471,7 +471,7 @@ 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 %d 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);
@@ -588,12 +588,12 @@ 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,

                                    audio_format_t format,

                                    uint32_t channels,

                                    audio_channel_mask_t channelMask,

                                    audio_output_flags_t flags)

{

    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();

    if (aps == 0) return 0;

    return aps->getOutput(stream, samplingRate, format, channels, flags);

    return aps->getOutput(stream, samplingRate, format, channelMask, flags);

}



status_t AudioSystem::startOutput(audio_io_handle_t output,
@@ -624,12 +624,12 @@ void AudioSystem::releaseOutput(audio_io_handle_t output) 
audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,

                                    uint32_t samplingRate,

                                    audio_format_t format,

                                    uint32_t channels,

                                    audio_channel_mask_t channelMask,

                                    int sessionId)

{

    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();

    if (aps == 0) return 0;

    return aps->getInput(inputSource, samplingRate, format, channels, sessionId);

    return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);

}



status_t AudioSystem::startInput(audio_io_handle_t input)

media/libmedia/IAudioPolicyService.cpp

+8 −8
Original line number Diff line number Diff line
@@ -123,7 +123,7 @@ public: 
                                        audio_stream_type_t stream,

                                        uint32_t samplingRate,

                                        audio_format_t format,

                                        uint32_t channels,

                                        audio_channel_mask_t channelMask,

                                        audio_output_flags_t flags)

    {

        Parcel data, reply;
@@ -131,7 +131,7 @@ public: 
        data.writeInt32(static_cast <uint32_t>(stream));

        data.writeInt32(samplingRate);

        data.writeInt32(static_cast <uint32_t>(format));

        data.writeInt32(channels);

        data.writeInt32(channelMask);

        data.writeInt32(static_cast <uint32_t>(flags));

        remote()->transact(GET_OUTPUT, data, &reply);

        return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -175,7 +175,7 @@ public: 
                                    audio_source_t inputSource,

                                    uint32_t samplingRate,

                                    audio_format_t format,

                                    uint32_t channels,

                                    audio_channel_mask_t channelMask,

                                    int audioSession)

    {

        Parcel data, reply;
@@ -183,7 +183,7 @@ public: 
        data.writeInt32((int32_t) inputSource);

        data.writeInt32(samplingRate);

        data.writeInt32(static_cast <uint32_t>(format));

        data.writeInt32(channels);

        data.writeInt32(channelMask);

        data.writeInt32(audioSession);

        remote()->transact(GET_INPUT, data, &reply);

        return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -415,14 +415,14 @@ status_t BnAudioPolicyService::onTransact( 
                    static_cast <audio_stream_type_t>(data.readInt32());

            uint32_t samplingRate = data.readInt32();

            audio_format_t format = (audio_format_t) data.readInt32();

            uint32_t channels = data.readInt32();

            audio_channel_mask_t channelMask = data.readInt32();

            audio_output_flags_t flags =

                    static_cast <audio_output_flags_t>(data.readInt32());



            audio_io_handle_t output = getOutput(stream,

                                                 samplingRate,

                                                 format,

                                                 channels,

                                                 channelMask,

                                                 flags);

            reply->writeInt32(static_cast <int>(output));

            return NO_ERROR;
@@ -462,12 +462,12 @@ status_t BnAudioPolicyService::onTransact( 
            audio_source_t inputSource = (audio_source_t) data.readInt32();

            uint32_t samplingRate = data.readInt32();

            audio_format_t format = (audio_format_t) data.readInt32();

            uint32_t channels = data.readInt32();

            audio_channel_mask_t channelMask = data.readInt32();

            int audioSession = data.readInt32();

            audio_io_handle_t input = getInput(inputSource,

                                               samplingRate,

                                               format,

                                               channels,

                                               channelMask,

                                               audioSession);

            reply->writeInt32(static_cast <int>(input));

            return NO_ERROR;

services/audioflinger/AudioFlinger.cpp

+21 −26
Original line number Diff line number Diff line
@@ -432,7 +432,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( 
        audio_stream_type_t streamType,

        uint32_t sampleRate,

        audio_format_t format,

        uint32_t channelMask,

        audio_channel_mask_t channelMask,

        int frameCount,

        IAudioFlinger::track_flags_t flags,

        const sp<IMemory>& sharedBuffer,
@@ -1652,7 +1652,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac 
        audio_stream_type_t streamType,

        uint32_t sampleRate,

        audio_format_t format,

        uint32_t channelMask,

        audio_channel_mask_t channelMask,

        int frameCount,

        const sp<IMemory>& sharedBuffer,

        int sessionId,
@@ -1707,7 +1707,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac 
                frameCount, mFrameCount);

      } else {

        ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "

                "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d "

                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d "

                "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",

                isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,

                audio_is_linear_pcm(format),
@@ -2192,7 +2192,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud 
        // mNormalSink below

{

    ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type);

    ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "

    ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "

            "mFrameCount=%d, mNormalFrameCount=%d",

            mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,

            mNormalFrameCount);
@@ -3466,7 +3466,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() 
                readOutputParameters();

                mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);

                for (size_t i = 0; i < mTracks.size() ; i++) {

                    int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask);

                    int name = getTrackName_l(mTracks[i]->mChannelMask);

                    if (name < 0) break;

                    mTracks[i]->mName = name;

                    // limit track sample rate to 2 x new output sample rate
@@ -4061,7 +4061,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( 
            const sp<Client>& client,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount,

            const sp<IMemory>& sharedBuffer,

            int sessionId)
@@ -4236,7 +4236,7 @@ AudioFlinger::PlaybackThread::Track::Track( 
            audio_stream_type_t streamType,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount,

            const sp<IMemory>& sharedBuffer,

            int sessionId,
@@ -4262,7 +4262,7 @@ AudioFlinger::PlaybackThread::Track::Track( 
        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack

        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);

        // to avoid leaking a track name, do not allocate one unless there is an mCblk

        mName = thread->getTrackName_l((audio_channel_mask_t)channelMask);

        mName = thread->getTrackName_l(channelMask);

        mCblk->mName = mName;

        if (mName < 0) {

            ALOGE("no more track names available");
@@ -4781,7 +4781,7 @@ AudioFlinger::PlaybackThread::TimedTrack::create( 
            audio_stream_type_t streamType,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount,

            const sp<IMemory>& sharedBuffer,

            int sessionId) {
@@ -4799,7 +4799,7 @@ AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 
            audio_stream_type_t streamType,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount,

            const sp<IMemory>& sharedBuffer,

            int sessionId)
@@ -5294,7 +5294,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( 
            const sp<Client>& client,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount,

            int sessionId)

    :   TrackBase(thread, client, sampleRate, format,
@@ -5406,7 +5406,7 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 
            DuplicatingThread *sourceThread,

            uint32_t sampleRate,

            audio_format_t format,

            uint32_t channelMask,

            audio_channel_mask_t channelMask,

            int frameCount)

    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,

                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
@@ -5794,7 +5794,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( 
        audio_io_handle_t input,

        uint32_t sampleRate,

        audio_format_t format,

        uint32_t channelMask,

        audio_channel_mask_t channelMask,

        int frameCount,

        IAudioFlinger::track_flags_t flags,

        int *sessionId,
@@ -5899,13 +5899,13 @@ status_t AudioFlinger::RecordHandle::onTransact( 
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,

                                         AudioStreamIn *input,

                                         uint32_t sampleRate,

                                         uint32_t channels,

                                         audio_channel_mask_t channelMask,

                                         audio_io_handle_t id,

                                         uint32_t device) :

    ThreadBase(audioFlinger, id, device, RECORD),

    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),

    // mRsmpInIndex and mInputBytes set by readInputParameters()

    mReqChannelCount(popcount(channels)),

    mReqChannelCount(popcount(channelMask)),

    mReqSampleRate(sampleRate)

    // mBytesRead is only meaningful while active, and so is cleared in start()

    // (but might be better to also clear here for dump?)
@@ -6149,7 +6149,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR 
        const sp<AudioFlinger::Client>& client,

        uint32_t sampleRate,

        audio_format_t format,

        int channelMask,

        audio_channel_mask_t channelMask,

        int frameCount,

        int sessionId,

        status_t *status)
@@ -6929,7 +6929,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 
                                          audio_devices_t *pDevices,

                                          uint32_t *pSamplingRate,

                                          audio_format_t *pFormat,

                                          uint32_t *pChannelMask)

                                          audio_channel_mask_t *pChannelMask)

{

    status_t status;

    RecordThread *thread = NULL;
@@ -6972,7 +6972,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 
        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&

        (config.sample_rate <= 2 * reqSamplingRate) &&

        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {

        ALOGV("openInput() reopening with proposed sampling rate and channels");

        ALOGV("openInput() reopening with proposed sampling rate and channel mask");

        inStream = NULL;

        status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream);

    }
@@ -8193,7 +8193,6 @@ void AudioFlinger::EffectModule::reset_l() 


status_t AudioFlinger::EffectModule::configure()

{

    uint32_t channels;

    if (mEffectInterface == NULL) {

        return NO_INIT;

    }
@@ -8204,18 +8203,14 @@ status_t AudioFlinger::EffectModule::configure() 
    }



    // TODO: handle configuration of effects replacing track process

    if (thread->channelCount() == 1) {

        channels = AUDIO_CHANNEL_OUT_MONO;

    } else {

        channels = AUDIO_CHANNEL_OUT_STEREO;

    }

    audio_channel_mask_t channelMask = thread->channelMask();



    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {

        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;

    } else {

        mConfig.inputCfg.channels = channels;

        mConfig.inputCfg.channels = channelMask;

    }

    mConfig.outputCfg.channels = channels;

    mConfig.outputCfg.channels = channelMask;

    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;

    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;

    mConfig.inputCfg.samplingRate = thread->sampleRate();