Audio笔记之重采样

AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
        Vector< sp<Track> > *tracksToRemove)
{
    mAudioMixer->setParameter(
                name,
                AudioMixer::RESAMPLE,
                AudioMixer::SAMPLE_RATE,
                (void *)(uintptr_t)reqSampleRate);
}

void AudioMixer::setParameter(int name, int target, int param, void *value)
{
    name -= TRACK0;
    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
    track_t& track = mState.tracks[name];

    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);

    switch (target) {
    ......
    case RESAMPLE:
        switch (param) {
        case SAMPLE_RATE:
            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
            if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
                        uint32_t(valueInt));
                invalidateState(1 << name);
            }
            break;
        case RESET:
            track.resetResampler();
            invalidateState(1 << name);
            break;
        case REMOVE:
            delete track.resampler;
            track.resampler = NULL;
            track.sampleRate = mSampleRate;
            invalidateState(1 << name);
            break;
        default:
            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
        }
        break;
    ......
    default:
        LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
    }
}

bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
{
    if (trackSampleRate != devSampleRate || resampler != NULL) {
        if (sampleRate != trackSampleRate) {
            sampleRate = trackSampleRate;
            if (resampler == NULL) {
                ALOGV("Creating resampler from track %d Hz to device %d Hz",
                        trackSampleRate, devSampleRate);
                AudioResampler::src_quality quality;
                // force lowest quality level resampler if use case isn't music or video
                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
                // quality level based on the initial ratio, but that could change later.
                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
                if (!((trackSampleRate == 44100 && devSampleRate == 48000) ||
                      (trackSampleRate == 48000 && devSampleRate == 44100))) {
                    quality = AudioResampler::DYN_LOW_QUALITY;
                } else {
                    quality = AudioResampler::DEFAULT_QUALITY;
                }

                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
                // but if none exists, it is the channel count (1 for mono).
                const int resamplerChannelCount = downmixerBufferProvider != NULL
                        ? mMixerChannelCount : channelCount;
                ALOGVV("Creating resampler:"
                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
                resampler = AudioResampler::create(
                        mMixerInFormat,
                        resamplerChannelCount,
                        devSampleRate, quality);
                resampler->setLocalTimeFreq(sLocalTimeFreq);
            }
            return true;
        }
    }
    return false;
}

void AudioFlinger::MixerThread::threadLoop_mix()
{
    // obtain the presentation timestamp of the next output buffer
    int64_t pts;
    status_t status = INVALID_OPERATION;

    if (mNormalSink != 0) {
        status = mNormalSink->getNextWriteTimestamp(&pts);
    } else {
        status = mOutputSink->getNextWriteTimestamp(&pts);
    }

    if (status != NO_ERROR) {
        pts = AudioBufferProvider::kInvalidPTS;
    }

    // mix buffers...
    mAudioMixer->process(pts);
    mCurrentWriteLength = mSinkBufferSize;
    // increase sleep time progressively when application underrun condition clears.
    // Only increase sleep time if the mixer is ready for two consecutive times to avoid
    // that a steady state of alternating ready/not ready conditions keeps the sleep time
    // such that we would underrun the audio HAL.
    if ((sleepTime == 0) && (sleepTimeShift > 0)) {
        sleepTimeShift--;
    }
    sleepTime = 0;
    standbyTime = systemTime() + standbyDelay;
    //TODO: delay standby when effects have a tail

}

<pre name="code" class="cpp">void AudioMixer::process(int64_t pts)
{
    mState.hook(&mState, pts);
}

void AudioMixer::invalidateState(uint32_t mask)
{
    if (mask != 0) {
        mState.needsChanged |= mask;
        mState.hook = process__validate;
    }
 }

bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
{
    if (trackSampleRate != devSampleRate || resampler != NULL) {
        if (sampleRate != trackSampleRate) {
            sampleRate = trackSampleRate;
            if (resampler == NULL) {
                ALOGV("Creating resampler from track %d Hz to device %d Hz",
                        trackSampleRate, devSampleRate);
                AudioResampler::src_quality quality;
                // force lowest quality level resampler if use case isn't music or video
                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
                // quality level based on the initial ratio, but that could change later.
                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
                if (!((trackSampleRate == 44100 && devSampleRate == 48000) ||
                      (trackSampleRate == 48000 && devSampleRate == 44100))) {
                    quality = AudioResampler::DYN_LOW_QUALITY;
                } else {
                    quality = AudioResampler::DEFAULT_QUALITY;
                }

                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
                // but if none exists, it is the channel count (1 for mono).
                const int resamplerChannelCount = downmixerBufferProvider != NULL
                        ? mMixerChannelCount : channelCount;
                ALOGVV("Creating resampler:"
                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
                resampler = AudioResampler::create(
                        mMixerInFormat,
                        resamplerChannelCount,
                        devSampleRate, quality);
                resampler->setLocalTimeFreq(sLocalTimeFreq);
            }
            return true;
        }
    }
    return false;
}

void AudioMixer::process__validate(state_t* state, int64_t pts)
{
    ALOGW_IF(!state->needsChanged,
        "in process__validate() but nothing's invalid");

    uint32_t changed = state->needsChanged;
    state->needsChanged = 0; // clear the validation flag

    // recompute which tracks are enabled / disabled
    uint32_t enabled = 0;
    uint32_t disabled = 0;
    while (changed) {
        const int i = 31 - __builtin_clz(changed);
        const uint32_t mask = 1<<i;
        changed &= ~mask;
        track_t& t = state->tracks[i];
        (t.enabled ? enabled : disabled) |= mask;
    }
    state->enabledTracks &= ~disabled;
    state->enabledTracks |=  enabled;

    // compute everything we need...
    int countActiveTracks = 0;
    // TODO: fix all16BitsStereNoResample logic to
    // either properly handle muted tracks (it should ignore them)
    // or remove altogether as an obsolete optimization.
    bool all16BitsStereoNoResample = true;
    bool resampling = false;
    bool volumeRamp = false;
    uint32_t en = state->enabledTracks;
    while (en) {
        const int i = 31 - __builtin_clz(en);
        en &= ~(1<<i);

        countActiveTracks++;
        track_t& t = state->tracks[i];
        uint32_t n = 0;
        // FIXME can overflow (mask is only 3 bits)
        n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
        if (t.doesResample()) {
            n |= NEEDS_RESAMPLE;
        }
        if (t.auxLevel != 0 && t.auxBuffer != NULL) {
            n |= NEEDS_AUX;
        }

        if (t.volumeInc[0]|t.volumeInc[1]) {
            volumeRamp = true;
        } else if (!t.doesResample() && t.volumeRL == 0) {
            n |= NEEDS_MUTE;
        }
        t.needs = n;

        if (n & NEEDS_MUTE) {
            t.hook = track__nop;
        } else {
            if (n & NEEDS_AUX) {
                all16BitsStereoNoResample = false;
            }
            if (n & NEEDS_RESAMPLE) {
                all16BitsStereoNoResample = false;
                resampling = true;
                t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
                        t.mMixerInFormat, t.mMixerFormat);
                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
                        "Track %d needs downmix + resample", i);
            } else {
                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
                    t.hook = getTrackHook(
                            t.mMixerChannelCount == 2 // TODO: MONO_HACK.
                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
                            t.mMixerChannelCount,
                            t.mMixerInFormat, t.mMixerFormat);
                    all16BitsStereoNoResample = false;
                }
                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
                    t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
                            t.mMixerInFormat, t.mMixerFormat);
                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
                            "Track %d needs downmix", i);
                }
            }
        }
    }

    // select the processing hooks
    state->hook = process__nop;
    if (countActiveTracks > 0) {
        if (resampling) {
            if (!state->outputTemp) {
                state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
            }
            if (!state->resampleTemp) {
                state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
            }
            state->hook = process__genericResampling;
        } else {
            if (state->outputTemp) {
                delete [] state->outputTemp;
                state->outputTemp = NULL;
            }
            if (state->resampleTemp) {
                delete [] state->resampleTemp;
                state->resampleTemp = NULL;
            }
            state->hook = process__genericNoResampling;
            if (all16BitsStereoNoResample && !volumeRamp) {
                if (countActiveTracks == 1) {
                    const int i = 31 - __builtin_clz(state->enabledTracks);
                    track_t& t = state->tracks[i];
                    if ((t.needs & NEEDS_MUTE) == 0) {
                        // The check prevents a muted track from acquiring a process hook.
                        //
                        // This is dangerous if the track is MONO as that requires
                        // special case handling due to implicit channel duplication.
                        // Stereo or Multichannel should actually be fine here.
                        state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
                                t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
                    }
                }
            }
        }
    }

    ALOGV("mixer configuration change: %d activeTracks (%08x) "
        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
        countActiveTracks, state->enabledTracks,
        all16BitsStereoNoResample, resampling, volumeRamp);

   state->hook(state, pts);

    // Now that the volume ramp has been done, set optimal state and
    // track hooks for subsequent mixer process
    if (countActiveTracks > 0) {
        bool allMuted = true;
        uint32_t en = state->enabledTracks;
        while (en) {
            const int i = 31 - __builtin_clz(en);
            en &= ~(1<<i);
            track_t& t = state->tracks[i];
            if (!t.doesResample() && t.volumeRL == 0) {
                t.needs |= NEEDS_MUTE;
                t.hook = track__nop;
            } else {
                allMuted = false;
            }
        }
        if (allMuted) {
            state->hook = process__nop;
        } else if (all16BitsStereoNoResample) {
            if (countActiveTracks == 1) {
                const int i = 31 - __builtin_clz(state->enabledTracks);
                track_t& t = state->tracks[i];
                // Muted single tracks handled by allMuted above.
                state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
                        t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
            }
        }
    }
}

void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
        int32_t* temp, int32_t* aux)
{
    ALOGVV("track__genericResample\n");
    t->resampler->setSampleRate(t->sampleRate);

    // ramp gain - resample to temp buffer and scale/mix in 2nd step
    if (aux != NULL) {
        // always resample with unity gain when sending to auxiliary buffer to be able
        // to apply send level after resampling
        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
        t->resampler->resample(temp, outFrameCount, t->bufferProvider);
        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
            volumeRampStereo(t, out, outFrameCount, temp, aux);
        } else {
            volumeStereo(t, out, outFrameCount, temp, aux);
        }
    } else {
        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
            t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
            t->resampler->resample(temp, outFrameCount, t->bufferProvider);
            volumeRampStereo(t, out, outFrameCount, temp, aux);
        }

        // constant gain
        else {
            t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
            t->resampler->resample(out, outFrameCount, t->bufferProvider);
        }
    }
}