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Commit a1ab7cc8 authored by Andy Hung's avatar Andy Hung
Browse files

Add Track Sink Format to AudioMixer 



Track Sink Format now allows both AUDIO_FORMAT_PCM_FLOAT and
AUDIO_FORMAT_PCM_16_BIT (default).  The float case is not
enabled now.

Change-Id: Ibee70cab4725c8bc9905e49f1f9a9c2448f76e00
Signed-off-by: default avatarAndy Hung <hunga@google.com>
parent 56df9ff3

services/audioflinger/AudioMixer.cpp

+68 −22
Original line number Diff line number Diff line
@@ -193,6 +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;



        status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);

        if (status == OK) {
@@ -440,6 +441,13 @@ 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: {

            audio_format_t format = static_cast<audio_format_t>(valueInt);

            if (track.mSinkFormat != format) {

                track.mSinkFormat = format;

                ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);

            }

            } break;

        default:

            LOG_FATAL("bad param");

        }
@@ -1043,7 +1051,7 @@ void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, 
void AudioMixer::process__nop(state_t* state, int64_t pts)

{

    uint32_t e0 = state->enabledTracks;

    size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;

    size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;

    while (e0) {

        // process by group of tracks with same output buffer to

        // avoid multiple memset() on same buffer
@@ -1062,7 +1070,8 @@ void AudioMixer::process__nop(state_t* state, int64_t pts) 
            }

            e0 &= ~(e1);



            memset(t1.mainBuffer, 0, bufSize);

            memset(t1.mainBuffer, 0, sampleCount

                    * audio_bytes_per_sample(t1.mSinkFormat));

        }



        while (e1) {
@@ -1170,8 +1179,18 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) 
                    }

                }

            }

            switch (t1.mSinkFormat) {

            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.

                break;

            case AUDIO_FORMAT_PCM_16_BIT:

                ditherAndClamp(out, outTemp, BLOCKSIZE);

            out += BLOCKSIZE;

                out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame

                break;

            default:

                LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);

            }

            numFrames += BLOCKSIZE;

        } while (numFrames < state->frameCount);

    }
@@ -1253,7 +1272,16 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) 
                }

            }

        }

        switch (t1.mSinkFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);

            break;

        case AUDIO_FORMAT_PCM_16_BIT:

            ditherAndClamp(out, outTemp, numFrames);

            break;

        default:

            LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);

        }

    }

}
@@ -1294,6 +1322,20 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, 
        }

        size_t outFrames = b.frameCount;



        switch (t.mSinkFormat) {

        case AUDIO_FORMAT_PCM_FLOAT: {

            float *fout = reinterpret_cast<float*>(out);

            static float scale = 1. / (32768. * 4096.); // exact when inverted

            do {

                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);

                in += 2;

                int32_t l = mulRL(1, rl, vrl);

                int32_t r = mulRL(0, rl, vrl);

                *fout++ = static_cast<float>(l) * scale;

                *fout++ = static_cast<float>(r) * scale;

            } while (--outFrames);

            } break;

        case AUDIO_FORMAT_PCM_16_BIT:

            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {

                // volume is boosted, so we might need to clamp even though

                // we process only one track.
@@ -1316,6 +1358,10 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, 
                    *out++ = (r<<16) | (l & 0xFFFF);

                } while (--outFrames);

            }

            break;

        default:

            LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);

        }

        numFrames -= b.frameCount;

        t.bufferProvider->releaseBuffer(&b);

    }

services/audioflinger/AudioMixer.h

+4 −1
Original line number Diff line number Diff line
@@ -77,6 +77,7 @@ public: 
        MAIN_BUFFER     = 0x4002,

        AUX_BUFFER      = 0x4003,

        DOWNMIX_TYPE    = 0X4004,

        SINK_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.
@@ -193,7 +194,9 @@ private: 


        int32_t     sessionId;



        int32_t     padding[2];

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



        int32_t     padding[1];



        // 16-byte boundary