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Commit 78820705 authored by Andy Hung's avatar Andy Hung
Browse files

Rename mSinkFormat to mMixerFormat for AudioMixer::track_t



AudioMixer::SINK_FORMAT also changes to AudioMixer::MIXER_FORMAT

Change-Id: Ic3f8be77d2c75c082c4fd140bc907e30c304d285
Signed-off-by: default avatarAndy Hung <hunga@google.com>
parent 7ed1873a
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+12 −12
Original line number Diff line number Diff line
@@ -193,7 +193,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
        t->mainBuffer = NULL;
        t->auxBuffer = NULL;
        t->downmixerBufferProvider = NULL;
        t->mSinkFormat = AUDIO_FORMAT_PCM_16_BIT;
        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;

        status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
        if (status == OK) {
@@ -441,11 +441,11 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
        //         for a specific track? or per mixer?
        /* case DOWNMIX_TYPE:
            break          */
        case SINK_FORMAT: {
        case MIXER_FORMAT: {
            audio_format_t format = static_cast<audio_format_t>(valueInt);
            if (track.mSinkFormat != format) {
                track.mSinkFormat = format;
                ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);
            if (track.mMixerFormat != format) {
                track.mMixerFormat = format;
                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
            }
            } break;
        default:
@@ -1071,7 +1071,7 @@ void AudioMixer::process__nop(state_t* state, int64_t pts)
            e0 &= ~(e1);

            memset(t1.mainBuffer, 0, sampleCount
                    * audio_bytes_per_sample(t1.mSinkFormat));
                    * audio_bytes_per_sample(t1.mMixerFormat));
        }

        while (e1) {
@@ -1179,7 +1179,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
                    }
                }
            }
            switch (t1.mSinkFormat) {
            switch (t1.mMixerFormat) {
            case AUDIO_FORMAT_PCM_FLOAT:
                memcpy_to_float_from_q19_12(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
                out += BLOCKSIZE * 2; // output is 2 floats/frame.
@@ -1189,7 +1189,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
                out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
                break;
            default:
                LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
                LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
            }
            numFrames += BLOCKSIZE;
        } while (numFrames < state->frameCount);
@@ -1272,7 +1272,7 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
                }
            }
        }
        switch (t1.mSinkFormat) {
        switch (t1.mMixerFormat) {
        case AUDIO_FORMAT_PCM_FLOAT:
            memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
            break;
@@ -1280,7 +1280,7 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
            ditherAndClamp(out, outTemp, numFrames);
            break;
        default:
            LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
            LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
        }
    }
}
@@ -1322,7 +1322,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
        }
        size_t outFrames = b.frameCount;

        switch (t.mSinkFormat) {
        switch (t.mMixerFormat) {
        case AUDIO_FORMAT_PCM_FLOAT: {
            float *fout = reinterpret_cast<float*>(out);
            do {
@@ -1361,7 +1361,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
            }
            break;
        default:
            LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);
            LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
        }
        numFrames -= b.frameCount;
        t.bufferProvider->releaseBuffer(&b);
+2 −2
Original line number Diff line number Diff line
@@ -77,7 +77,7 @@ public:
        MAIN_BUFFER     = 0x4002,
        AUX_BUFFER      = 0x4003,
        DOWNMIX_TYPE    = 0X4004,
        SINK_FORMAT     = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
        // for target RESAMPLE
        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
                                  // parameter 'value' is the new sample rate in Hz.
@@ -194,7 +194,7 @@ private:

        int32_t     sessionId;

        audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
        audio_format_t mMixerFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)

        int32_t     padding[1];

+2 −2
Original line number Diff line number Diff line
@@ -3293,7 +3293,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
                        AudioMixer::SINK_FORMAT, (void *)mMixerBufferFormat);
                        AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
@@ -3304,7 +3304,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,
                        AudioMixer::SINK_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
                        AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
                mAudioMixer->setParameter(
                        name,
                        AudioMixer::TRACK,