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Commit 043bdaaf authored by Andy Hung's avatar Andy Hung Committed by Android Git Automerger
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am 7a9b2594: Merge "Remove unused TwoTrack process mode from AudioMixer" into lmp-dev

* commit '7a9b2594':
  Remove unused TwoTrack process mode from AudioMixer
parents ed977de0 7a9b2594
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+0 −103
Original line number Diff line number Diff line
@@ -1797,109 +1797,6 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
    }
}

#if 0
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
                                                            int64_t pts)
{
    int i;
    uint32_t en = state->enabledTracks;

    i = 31 - __builtin_clz(en);
    const track_t& t0 = state->tracks[i];
    AudioBufferProvider::Buffer& b0(t0.buffer);

    en &= ~(1<<i);
    i = 31 - __builtin_clz(en);
    const track_t& t1 = state->tracks[i];
    AudioBufferProvider::Buffer& b1(t1.buffer);

    const int16_t *in0;
    const int16_t vl0 = t0.volume[0];
    const int16_t vr0 = t0.volume[1];
    size_t frameCount0 = 0;

    const int16_t *in1;
    const int16_t vl1 = t1.volume[0];
    const int16_t vr1 = t1.volume[1];
    size_t frameCount1 = 0;

    //FIXME: only works if two tracks use same buffer
    int32_t* out = t0.mainBuffer;
    size_t numFrames = state->frameCount;
    const int16_t *buff = NULL;


    while (numFrames) {

        if (frameCount0 == 0) {
            b0.frameCount = numFrames;
            int64_t outputPTS = calculateOutputPTS(t0, pts,
                                                   out - t0.mainBuffer);
            t0.bufferProvider->getNextBuffer(&b0, outputPTS);
            if (b0.i16 == NULL) {
                if (buff == NULL) {
                    buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
                }
                in0 = buff;
                b0.frameCount = numFrames;
            } else {
                in0 = b0.i16;
            }
            frameCount0 = b0.frameCount;
        }
        if (frameCount1 == 0) {
            b1.frameCount = numFrames;
            int64_t outputPTS = calculateOutputPTS(t1, pts,
                                                   out - t0.mainBuffer);
            t1.bufferProvider->getNextBuffer(&b1, outputPTS);
            if (b1.i16 == NULL) {
                if (buff == NULL) {
                    buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
                }
                in1 = buff;
                b1.frameCount = numFrames;
            } else {
                in1 = b1.i16;
            }
            frameCount1 = b1.frameCount;
        }

        size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1;

        numFrames -= outFrames;
        frameCount0 -= outFrames;
        frameCount1 -= outFrames;

        do {
            int32_t l0 = *in0++;
            int32_t r0 = *in0++;
            l0 = mul(l0, vl0);
            r0 = mul(r0, vr0);
            int32_t l = *in1++;
            int32_t r = *in1++;
            l = mulAdd(l, vl1, l0) >> 12;
            r = mulAdd(r, vr1, r0) >> 12;
            // clamping...
            l = clamp16(l);
            r = clamp16(r);
            *out++ = (r<<16) | (l & 0xFFFF);
        } while (--outFrames);

        if (frameCount0 == 0) {
            t0.bufferProvider->releaseBuffer(&b0);
        }
        if (frameCount1 == 0) {
            t1.bufferProvider->releaseBuffer(&b1);
        }
    }

    delete [] buff;
}
#endif

int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
                                       int outputFrameIndex)
{
+0 −4
Original line number Diff line number Diff line
@@ -408,10 +408,6 @@ private:
    static void process__genericResampling(state_t* state, int64_t pts);
    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
                                                          int64_t pts);
#if 0
    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
                                                           int64_t pts);
#endif

    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
                                      int outputFrameIndex);