C++ AudioBuffer::channels方法代码示例

// This one puts some data inside the ring buffer.
void RingBuffer::put(AudioBuffer& buf)
{
    std::lock_guard<std::mutex> l(lock_);
    const size_t sample_num = buf.frames();
    const size_t buffer_size = buffer_.frames();
    if (buffer_size == 0)
        return;

    size_t len = putLength();
    if (buffer_size - len < sample_num)
        discard(sample_num);
    size_t toCopy = sample_num;

    // Add more channels if the input buffer holds more channels than the ring.
    if (buffer_.channels() < buf.channels())
        buffer_.setChannelNum(buf.channels());

    size_t in_pos = 0;
    size_t pos = endPos_;

    while (toCopy) {
        size_t block = toCopy;

        if (block > buffer_size - pos) // Wrap block around ring ?
            block = buffer_size - pos; // Fill in to the end of the buffer

        buffer_.copy(buf, block, in_pos, pos);
        in_pos += block;
        pos = (pos + block) % buffer_size;
        toCopy -= block;
    }

    endPos_ = pos;
    not_empty_.notify_all();
}

void Resampler::resample(const AudioBuffer &dataIn, AudioBuffer &dataOut)
{
    const double inputFreq = dataIn.getSampleRate();
    const double outputFreq = dataOut.getSampleRate();
    const double sampleFactor = outputFreq / inputFreq;

    if (sampleFactor == 1.0)
        return;

    const size_t nbFrames = dataIn.frames();
    const size_t nbChans = dataIn.channels();

    if (nbChans != format_.nb_channels) {
        // change channel num if needed
        int err;
        src_delete(src_state_);
        src_state_ = src_new(SRC_LINEAR, nbChans, &err);
        format_.nb_channels = nbChans;
        DEBUG("SRC channel number changed.");
    }
    if (nbChans != dataOut.channels()) {
        DEBUG("Output buffer had the wrong number of channels (in: %d, out: %d).", nbChans, dataOut.channels());
        dataOut.setChannelNum(nbChans);
    }

    size_t inSamples = nbChans * nbFrames;
    size_t outSamples = inSamples * sampleFactor;

    // grow buffer if needed
    floatBufferIn_.resize(inSamples);
    floatBufferOut_.resize(outSamples);
    scratchBuffer_.resize(outSamples);

    SRC_DATA src_data;
    src_data.data_in = floatBufferIn_.data();
    src_data.data_out = floatBufferOut_.data();
    src_data.input_frames = nbFrames;
    src_data.output_frames = nbFrames * sampleFactor;
    src_data.src_ratio = sampleFactor;
    src_data.end_of_input = 0; // More data will come

    dataIn.interleaveFloat(floatBufferIn_.data());

    src_process(src_state_, &src_data);

    /*
    TODO: one-shot deinterleave and float-to-short conversion
    */
    src_float_to_short_array(floatBufferOut_.data(), scratchBuffer_.data(), outSamples);
    dataOut.deinterleave(scratchBuffer_.data(), src_data.output_frames, nbChans);
}

void AudioBuffer::convert_channels(AudioBuffer &_dest, unsigned _frames_count)
{
	AudioSpec destspec{m_spec.format, _dest.channels(), m_spec.rate};
	if(_dest.spec() != destspec) {
		throw std::logic_error("unsupported format");
	}

	_frames_count = std::min(frames(),_frames_count);
	if(m_spec.channels == destspec.channels) {
		_dest.add_frames(*this,_frames_count);
		return;
	}

	switch(m_spec.format) {
		case AUDIO_FORMAT_U8:
			convert_channels<uint8_t>(*this,_dest,_frames_count);
			break;
		case AUDIO_FORMAT_S16:
			convert_channels<int16_t>(*this,_dest,_frames_count);
			break;
		case AUDIO_FORMAT_F32:
			convert_channels<float>(*this,_dest,_frames_count);
			break;
		default:
			throw std::logic_error("unsupported format");
	}
}

void AudioBuffer::convert(const AudioSpec &_new_spec)
{
	if(_new_spec == m_spec) {
		return;
	}

	AudioSpec new_spec = _new_spec;
	AudioBuffer dest[2];
	unsigned bufidx = 0;
	AudioBuffer *source = this;

	if(source->rate() != new_spec.rate) {
#if HAVE_LIBSAMPLERATE
		if(source->format() != AUDIO_FORMAT_F32) {
			dest[1].set_spec({AUDIO_FORMAT_F32, source->channels(), source->rate()});
			source->convert_format(dest[1], source->frames());
			source = &dest[1];
		}
		dest[0].set_spec({source->format(), source->channels(), new_spec.rate});
		source->convert_rate(dest[0], source->frames(), nullptr);
		source = &dest[0];
		bufidx = 1;
#else
		new_spec.rate = source->rate();
#endif
	}
	if(source->channels() != new_spec.channels) {
		dest[bufidx].set_spec({source->format(),new_spec.channels,source->rate()});
		source->convert_channels(dest[bufidx], source->frames());
		source = &dest[bufidx];
		bufidx = (bufidx + 1) % 2;
	}
	if(source->format() != new_spec.format) {
		dest[bufidx].set_spec({new_spec.format,source->channels(),source->rate()});
		source->convert_format(dest[bufidx], source->frames());
		source = &dest[bufidx];
	}

	if(new_spec != m_spec) {
		m_data = source->m_data;
		m_spec = new_spec;
	}
}

void
JackLayer::write(AudioBuffer &buffer, std::vector<float> &floatBuffer)
{
    for (unsigned i = 0; i < out_ringbuffers_.size(); ++i) {
        const unsigned inChannel = std::min(i, buffer.channels() - 1);
        convertToFloat(*buffer.getChannel(inChannel), floatBuffer);

        // write to output
        const size_t to_ringbuffer = jack_ringbuffer_write_space(out_ringbuffers_[i]);
        const size_t write_bytes = std::min(buffer.frames() * sizeof(floatBuffer[0]), to_ringbuffer);
        // FIXME: while we have samples to write AND while we have space to write them
        const size_t written_bytes = jack_ringbuffer_write(out_ringbuffers_[i],
                (const char *) floatBuffer.data(), write_bytes);
        if (written_bytes < write_bytes)
            RING_WARN("Dropped %zu bytes for channel %u", write_bytes - written_bytes, i);
    }
}