C++ AudioBuffer::resize方法代码示例

void
JackLayer::read(AudioBuffer &buffer)
{
    for (unsigned i = 0; i < in_ringbuffers_.size(); ++i) {

        const size_t incomingSamples = jack_ringbuffer_read_space(in_ringbuffers_[i]) / sizeof(captureFloatBuffer_[0]);
        if (!incomingSamples)
            continue;

        captureFloatBuffer_.resize(incomingSamples);
        buffer.resize(incomingSamples);

        // write to output
        const size_t from_ringbuffer = jack_ringbuffer_read_space(in_ringbuffers_[i]);
        const size_t expected_bytes = std::min(incomingSamples * sizeof(captureFloatBuffer_[0]), from_ringbuffer);
        // FIXME: while we have samples to write AND while we have space to write them
        const size_t read_bytes = jack_ringbuffer_read(in_ringbuffers_[i],
                (char *) captureFloatBuffer_.data(), expected_bytes);
        if (read_bytes < expected_bytes) {
            RING_WARN("Dropped %zu bytes", expected_bytes - read_bytes);
            break;
        }

        /* Write the data one frame at a time.  This is
         * inefficient, but makes things simpler. */
        // FIXME: this is braindead, we should write blocks of samples at a time
        // convert a vector of samples from 1 channel to a float vector
        convertFromFloat(captureFloatBuffer_, *buffer.getChannel(i));
    }
}

void
AudioSender::process()
{
    auto& mainBuffer = Manager::instance().getRingBufferPool();
    auto mainBuffFormat = mainBuffer.getInternalAudioFormat();

    // compute nb of byte to get corresponding to 1 audio frame
    const std::size_t samplesToGet = std::chrono::duration_cast<std::chrono::seconds>(mainBuffFormat.sample_rate * secondsPerPacket_).count();

    if (mainBuffer.availableForGet(id_) < samplesToGet) {
        const auto wait_time = std::chrono::duration_cast<std::chrono::milliseconds>(secondsPerPacket_);
        if (not mainBuffer.waitForDataAvailable(id_, samplesToGet, wait_time))
            return;
    }

    // get data
    micData_.setFormat(mainBuffFormat);
    micData_.resize(samplesToGet);
    const auto samples = mainBuffer.getData(micData_, id_);
    if (samples != samplesToGet)
        return;

    // down/upmix as needed
    auto accountAudioCodec = std::static_pointer_cast<AccountAudioCodecInfo>(args_.codec);
    micData_.setChannelNum(accountAudioCodec->audioformat.nb_channels, true);

    if (mainBuffFormat.sample_rate != accountAudioCodec->audioformat.sample_rate) {
        if (not resampler_) {
            RING_DBG("Creating audio resampler");
            resampler_.reset(new Resampler(accountAudioCodec->audioformat));
        }
        resampledData_.setFormat(accountAudioCodec->audioformat);
        resampledData_.resize(samplesToGet);
        resampler_->resample(micData_, resampledData_);
        Smartools::getInstance().setLocalAudioCodec(audioEncoder_->getEncoderName());
        if (audioEncoder_->encode_audio(resampledData_) < 0)
            RING_ERR("encoding failed");
    } else {
        if (audioEncoder_->encode_audio(micData_) < 0)
            RING_ERR("encoding failed");
    }
}

void JackLayer::fillWithUrgent(AudioBuffer &buffer, size_t samplesToGet)
{
    // Urgent data (dtmf, incoming call signal) come first.
    samplesToGet = std::min(samplesToGet, hardwareBufferSize_);
    buffer.resize(samplesToGet);
    urgentRingBuffer_.get(buffer, RingBufferPool::DEFAULT_ID);
    buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);

    // Consume the regular one as well (same amount of samples)
    Manager::instance().getRingBufferPool().discard(samplesToGet, RingBufferPool::DEFAULT_ID);
}

size_t OpenSLLayer::audioPlaybackFillWithVoice(AudioBuffer &buffer)
{
    RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();
    size_t got = mainBuffer.getAvailableData(buffer, RingBufferPool::DEFAULT_ID);
    buffer.resize(got);
    buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
    if (audioFormat_.sample_rate != mainBuffer.getInternalSamplingRate()) {
        DEBUG("OpenSLLayer::audioPlaybackFillWithVoice sample_rate != mainBuffer.getInternalSamplingRate() \n");
        AudioBuffer out(buffer, false);
        out.setSampleRate(audioFormat_.sample_rate);
        resampler_->resample(buffer, out);
        buffer = out;
    }
    return buffer.size();
}

void JackLayer::fillWithVoice(AudioBuffer &buffer, size_t samplesAvail)
{
    RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();

    buffer.resize(samplesAvail);
    mainBuffer.getData(buffer, RingBufferPool::DEFAULT_ID);
    buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);

    if (audioFormat_.sample_rate != (unsigned) mainBuffer.getInternalSamplingRate()) {
        RING_DBG("fillWithVoice sample_rate != mainBuffer.getInternalSamplingRate() \n");
        AudioBuffer out(buffer, false);
        out.setSampleRate(audioFormat_.sample_rate);
        resampler_->resample(buffer, out);
        buffer = out;
    }
}

void JackLayer::fillWithToneOrRingtone(AudioBuffer &buffer)
{
    buffer.resize(hardwareBufferSize_);
    AudioLoop *tone = Manager::instance().getTelephoneTone();
    AudioLoop *file_tone = Manager::instance().getTelephoneFile();

    // In case of a dtmf, the pointers will be set to nullptr once the dtmf length is
    // reached. For this reason we need to fill audio buffer with zeros if pointer is nullptr
    if (tone) {
        tone->getNext(buffer, playbackGain_);
    } else if (file_tone) {
        file_tone->getNext(buffer, playbackGain_);
    } else {
        buffer.reset();
    }
}

size_t
RingBufferPool::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
{
    std::lock_guard<std::recursive_mutex> lk(stateLock_);

    auto bindings = getReadBindings(call_id);
    if (not bindings)
        return 0;

    // No mixing
    if (bindings->size() == 1) {
        return (*bindings->cbegin())->get(buffer, call_id);
    }

    size_t availableSamples = std::numeric_limits<size_t>::max();

    for (const auto& rbuf : *bindings)
        availableSamples = std::min(availableSamples,
                                    rbuf->availableForGet(call_id));

    if (availableSamples == std::numeric_limits<size_t>::max())
        return 0;

    availableSamples = std::min(availableSamples, buffer.frames());

    buffer.resize(availableSamples);
    buffer.reset();
    buffer.setFormat(internalAudioFormat_);

    AudioBuffer mixBuffer(buffer);

    for (const auto &rbuf : *bindings) {
        if (rbuf->get(mixBuffer, call_id) > 0)
            buffer.mix(mixBuffer);
    }

    return availableSamples;
}