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// Copyright (C) 2010-2016 Lukas Lalinsky
// Distributed under the MIT license, see the LICENSE file for details.
#include <assert.h>
#include <algorithm>
#include <stdio.h>
extern "C" {
#include "avresample/avcodec.h"
}
#include "debug.h"
#include "audio_processor.h"
namespace chromaprint {
static const int kMinSampleRate = 1000;
static const int kMaxBufferSize = 1024 * 32;
// Resampler configuration
static const int kResampleFilterLength = 16;
static const int kResamplePhaseShift = 8;
static const int kResampleLinear = 0;
static const double kResampleCutoff = 0.8;
AudioProcessor::AudioProcessor(int sample_rate, AudioConsumer *consumer)
: m_buffer(kMaxBufferSize),
m_buffer_offset(0),
m_resample_buffer(kMaxBufferSize),
m_target_sample_rate(sample_rate),
m_consumer(consumer),
m_resample_ctx(0)
{
}
AudioProcessor::~AudioProcessor()
{
if (m_resample_ctx) {
av_resample_close(m_resample_ctx);
}
}
void AudioProcessor::LoadMono(const int16_t *input, int length)
{
int16_t *output = m_buffer.data() + m_buffer_offset;
while (length--) {
*output++ = input[0];
input++;
}
}
void AudioProcessor::LoadStereo(const int16_t *input, int length)
{
int16_t *output = m_buffer.data() + m_buffer_offset;
while (length--) {
*output++ = (input[0] + input[1]) / 2;
input += 2;
}
}
void AudioProcessor::LoadMultiChannel(const int16_t *input, int length)
{
int16_t *output = m_buffer.data() + m_buffer_offset;
while (length--) {
int32_t sum = 0;
for (int i = 0; i < m_num_channels; i++) {
sum += *input++;
}
*output++ = (int16_t)(sum / m_num_channels);
}
}
int AudioProcessor::Load(const int16_t *input, int length)
{
assert(length >= 0);
assert(m_buffer_offset <= m_buffer.size());
length = std::min(length, static_cast<int>(m_buffer.size() - m_buffer_offset));
switch (m_num_channels) {
case 1:
LoadMono(input, length);
break;
case 2:
LoadStereo(input, length);
break;
default:
LoadMultiChannel(input, length);
break;
}
m_buffer_offset += length;
return length;
}
void AudioProcessor::Resample()
{
if (!m_resample_ctx) {
m_consumer->Consume(m_buffer.data(), m_buffer_offset);
m_buffer_offset = 0;
return;
}
int consumed = 0;
int length = av_resample(m_resample_ctx, m_resample_buffer.data(), m_buffer.data(), &consumed, m_buffer_offset, kMaxBufferSize, 1);
if (length > kMaxBufferSize) {
DEBUG("chromaprint::AudioProcessor::Resample() -- Resampling overwrote output buffer.");
length = kMaxBufferSize;
}
m_consumer->Consume(m_resample_buffer.data(), length);
int remaining = m_buffer_offset - consumed;
if (remaining > 0) {
std::copy(m_buffer.begin() + consumed, m_buffer.begin() + m_buffer_offset, m_buffer.begin());
}
else if (remaining < 0) {
DEBUG("chromaprint::AudioProcessor::Resample() -- Resampling overread input buffer.");
remaining = 0;
}
m_buffer_offset = remaining;
}
bool AudioProcessor::Reset(int sample_rate, int num_channels)
{
if (num_channels <= 0) {
DEBUG("chromaprint::AudioProcessor::Reset() -- No audio channels.");
return false;
}
if (sample_rate <= kMinSampleRate) {
DEBUG("chromaprint::AudioProcessor::Reset() -- Sample rate less than "
<< kMinSampleRate << " (" << sample_rate << ").");
return false;
}
m_buffer_offset = 0;
if (m_resample_ctx) {
av_resample_close(m_resample_ctx);
m_resample_ctx = 0;
}
if (sample_rate != m_target_sample_rate) {
m_resample_ctx = av_resample_init(
m_target_sample_rate, sample_rate,
kResampleFilterLength,
kResamplePhaseShift,
kResampleLinear,
kResampleCutoff);
}
m_num_channels = num_channels;
return true;
}
void AudioProcessor::Consume(const int16_t *input, int length)
{
assert(length >= 0);
assert(length % m_num_channels == 0);
length /= m_num_channels;
while (length > 0) {
int consumed = Load(input, length);
input += consumed * m_num_channels;
length -= consumed;
if (m_buffer.size() == m_buffer_offset) {
Resample();
if (m_buffer.size() == m_buffer_offset) {
DEBUG("chromaprint::AudioProcessor::Consume() -- Resampling failed?");
return;
}
}
}
}
void AudioProcessor::Flush()
{
if (m_buffer_offset) {
Resample();
}
}
}; // namespace chromaprint
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