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#include "audio_sampler.h"
#include <c10/util/Logging.h>
#include "util.h"
#define AVRESAMPLE_MAX_CHANNELS 32
// www.ffmpeg.org/doxygen/1.1/doc_2examples_2resampling_audio_8c-example.html#a24
namespace ffmpeg {
namespace {
int preparePlanes(
const AudioFormat& fmt,
const uint8_t* buffer,
int numSamples,
uint8_t** planes) {
int result;
if ((result = av_samples_fill_arrays(
planes,
nullptr, // linesize is not needed
buffer,
fmt.channels,
numSamples,
(AVSampleFormat)fmt.format,
1)) < 0) {
LOG(ERROR) << "av_samples_fill_arrays failed, err: "
<< Util::generateErrorDesc(result)
<< ", numSamples: " << numSamples << ", fmt: " << fmt.format;
}
return result;
}
} // namespace
AudioSampler::AudioSampler(void* logCtx) : logCtx_(logCtx) {}
AudioSampler::~AudioSampler() {
cleanUp();
}
void AudioSampler::shutdown() {
cleanUp();
}
bool AudioSampler::init(const SamplerParameters& params) {
cleanUp();
if (params.type != MediaType::TYPE_AUDIO) {
LOG(ERROR) << "Invalid media type, expected MediaType::TYPE_AUDIO";
return false;
}
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 28, 100)
SwrContext* swrContext_ = NULL;
AVChannelLayout channel_out;
AVChannelLayout channel_in;
av_channel_layout_default(&channel_out, params.out.audio.channels);
av_channel_layout_default(&channel_in, params.in.audio.channels);
int ret = swr_alloc_set_opts2(
&swrContext_,
&channel_out,
(AVSampleFormat)params.out.audio.format,
params.out.audio.samples,
&channel_in,
(AVSampleFormat)params.in.audio.format,
params.in.audio.samples,
0,
logCtx_);
#else
swrContext_ = swr_alloc_set_opts(
nullptr,
av_get_default_channel_layout(params.out.audio.channels),
(AVSampleFormat)params.out.audio.format,
params.out.audio.samples,
av_get_default_channel_layout(params.in.audio.channels),
(AVSampleFormat)params.in.audio.format,
params.in.audio.samples,
0,
logCtx_);
#endif
if (swrContext_ == nullptr) {
LOG(ERROR) << "Cannot allocate SwrContext";
return false;
}
int result;
if ((result = swr_init(swrContext_)) < 0) {
LOG(ERROR) << "swr_init failed, err: " << Util::generateErrorDesc(result)
<< ", in -> format: " << params.in.audio.format
<< ", channels: " << params.in.audio.channels
<< ", samples: " << params.in.audio.samples
<< ", out -> format: " << params.out.audio.format
<< ", channels: " << params.out.audio.channels
<< ", samples: " << params.out.audio.samples;
return false;
}
// set formats
params_ = params;
return true;
}
int AudioSampler::numOutputSamples(int inSamples) const {
return swr_get_out_samples(swrContext_, inSamples);
}
int AudioSampler::sample(
const uint8_t* inPlanes[],
int inNumSamples,
ByteStorage* out,
int outNumSamples) {
int result;
int outBufferBytes = av_samples_get_buffer_size(
nullptr,
params_.out.audio.channels,
outNumSamples,
(AVSampleFormat)params_.out.audio.format,
1);
if (out) {
out->ensure(outBufferBytes);
uint8_t* outPlanes[AVRESAMPLE_MAX_CHANNELS] = {nullptr};
if ((result = preparePlanes(
params_.out.audio,
out->writableTail(),
outNumSamples,
outPlanes)) < 0) {
return result;
}
if ((result = swr_convert(
swrContext_,
&outPlanes[0],
outNumSamples,
inPlanes,
inNumSamples)) < 0) {
LOG(ERROR) << "swr_convert failed, err: "
<< Util::generateErrorDesc(result);
return result;
}
TORCH_CHECK_LE(result, outNumSamples);
if (result) {
if ((result = av_samples_get_buffer_size(
nullptr,
params_.out.audio.channels,
result,
(AVSampleFormat)params_.out.audio.format,
1)) >= 0) {
out->append(result);
} else {
LOG(ERROR) << "av_samples_get_buffer_size failed, err: "
<< Util::generateErrorDesc(result);
}
}
} else {
// allocate a temporary buffer
auto* tmpBuffer = static_cast<uint8_t*>(av_malloc(outBufferBytes));
if (!tmpBuffer) {
LOG(ERROR) << "av_alloc failed, for size: " << outBufferBytes;
return -1;
}
uint8_t* outPlanes[AVRESAMPLE_MAX_CHANNELS] = {nullptr};
if ((result = preparePlanes(
params_.out.audio, tmpBuffer, outNumSamples, outPlanes)) < 0) {
av_free(tmpBuffer);
return result;
}
if ((result = swr_convert(
swrContext_,
&outPlanes[0],
outNumSamples,
inPlanes,
inNumSamples)) < 0) {
LOG(ERROR) << "swr_convert failed, err: "
<< Util::generateErrorDesc(result);
av_free(tmpBuffer);
return result;
}
av_free(tmpBuffer);
TORCH_CHECK_LE(result, outNumSamples);
if (result) {
result = av_samples_get_buffer_size(
nullptr,
params_.out.audio.channels,
result,
(AVSampleFormat)params_.out.audio.format,
1);
}
}
return result;
}
int AudioSampler::sample(AVFrame* frame, ByteStorage* out) {
const auto outNumSamples = numOutputSamples(frame ? frame->nb_samples : 0);
if (!outNumSamples) {
return 0;
}
return sample(
frame ? (const uint8_t**)&frame->data[0] : nullptr,
frame ? frame->nb_samples : 0,
out,
outNumSamples);
}
int AudioSampler::sample(const ByteStorage* in, ByteStorage* out) {
const auto inSampleSize =
av_get_bytes_per_sample((AVSampleFormat)params_.in.audio.format);
const auto inNumSamples =
!in ? 0 : in->length() / inSampleSize / params_.in.audio.channels;
const auto outNumSamples = numOutputSamples(inNumSamples);
if (!outNumSamples) {
return 0;
}
uint8_t* inPlanes[AVRESAMPLE_MAX_CHANNELS] = {nullptr};
int result;
if (in &&
(result = preparePlanes(
params_.in.audio, in->data(), inNumSamples, inPlanes)) < 0) {
return result;
}
return sample(
in ? (const uint8_t**)inPlanes : nullptr,
inNumSamples,
out,
outNumSamples);
}
void AudioSampler::cleanUp() {
if (swrContext_) {
swr_free(&swrContext_);
swrContext_ = nullptr;
}
}
} // namespace ffmpeg
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