std::uint8_t*
AudioDecoderFfmpeg::decodeFrame(const std::uint8_t* input,
std::uint32_t inputSize, std::uint32_t& outputSize)
{
//GNASH_REPORT_FUNCTION;
assert(inputSize);
size_t outSize = MAX_AUDIO_FRAME_SIZE;
// TODO: make this a private member, to reuse (see NetStreamFfmpeg in 0.8.3)
std::unique_ptr<std::int16_t, decltype(av_free)*> output(
reinterpret_cast<std::int16_t*>(av_malloc(outSize)), av_free );
if (!output.get()) {
log_error(_("failed to allocate audio buffer."));
outputSize = 0;
return nullptr;
}
std::int16_t* outPtr = output.get();
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("AudioDecoderFfmpeg: about to decode %d bytes; "
"ctx->channels:%d, ctx->frame_size:%d",
inputSize, _audioCodecCtx->channels, _audioCodecCtx->frame_size);
#endif
// older ffmpeg versions didn't accept a const input..
AVPacket pkt;
int got_frm = 0;
av_init_packet(&pkt);
pkt.data = const_cast<uint8_t*>(input);
pkt.size = inputSize;
std::unique_ptr<AVFrame, FrameDeleter> frm(FRAMEALLOC(), FrameDeleter());
if (!frm.get()) {
log_error(_("failed to allocate frame."));
return nullptr;
}
int tmp = avcodec_decode_audio4(_audioCodecCtx, frm.get(), &got_frm, &pkt);
#ifdef GNASH_DEBUG_AUDIO_DECODING
const char* fmtname = av_get_sample_fmt_name(_audioCodecCtx->sample_fmt);
log_debug(" decodeFrame | frm->nb_samples: %d | &got_frm: %d | "
"returned %d | inputSize: %d",
frm->nb_samples, got_frm, tmp, inputSize);
#endif
int plane_size;
if (tmp >= 0 && got_frm) {
int data_size = av_samples_get_buffer_size( &plane_size,
_audioCodecCtx->channels, frm->nb_samples,
_audioCodecCtx->sample_fmt, 1);
if (static_cast<int>(outSize) < data_size) {
log_error(_("output buffer size is too small for the current frame "
"(%d < %d)"), outSize, data_size);
return nullptr;
}
memcpy(outPtr, frm->extended_data[0], plane_size);
#if !(defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H))
int planar = av_sample_fmt_is_planar(_audioCodecCtx->sample_fmt);
if (planar && _audioCodecCtx->channels > 1) {
uint8_t *out = ((uint8_t *)outPtr) + plane_size;
for (int ch = 1; ch < _audioCodecCtx->channels; ch++) {
memcpy(out, frm->extended_data[ch], plane_size);
out += plane_size;
}
}
#endif
outSize = data_size;
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug(" decodeFrame | fmt: %d | fmt_name: %s | planar: %d | "
"plane_size: %d | outSize: %d",
_audioCodecCtx->sample_fmt, fmtname, planar, plane_size, outSize);
#endif
} else {
if (tmp < 0)
log_error(_("avcodec_decode_audio returned %d."), tmp);
if (outSize < 2)
log_error(_("outputSize:%d after decoding %d bytes of input audio "
"data."), outputSize, inputSize);
log_error(_("Upgrading ffmpeg/libavcodec might fix this issue."));
outputSize = 0;
return nullptr;
}
// Resampling is needed.
if (_resampler.init(_audioCodecCtx)) {
// Resampling is needed.
// Compute new size based on frame_size and
// resampling configuration
// Find out the needed sample rate scaling
double resampleFactor = 44100.0/_audioCodecCtx->sample_rate;
// Compute total number of input samples
int inSamples = outSize;
bool stereo = _audioCodecCtx->channels > 1 ? true : false;
if (stereo) inSamples = inSamples >> 1;
if (_audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16 ||
_audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16P) {
inSamples = inSamples >> 1;
}
std::uint8_t*
AudioDecoderFfmpeg::decodeFrame(const std::uint8_t* input,
std::uint32_t inputSize, std::uint32_t& outputSize)
{
//GNASH_REPORT_FUNCTION;
assert(inputSize);
size_t outSize = MAX_AUDIO_FRAME_SIZE;
// TODO: make this a private member, to reuse (see NetStreamFfmpeg in 0.8.3)
std::unique_ptr<std::int16_t, decltype(av_free)*> output(
reinterpret_cast<std::int16_t*>(av_malloc(outSize)), av_free );
if (!output.get()) {
log_error(_("failed to allocate audio buffer."));
outputSize = 0;
return nullptr;
}
std::int16_t* outPtr = output.get();
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("AudioDecoderFfmpeg: about to decode %d bytes; "
"ctx->channels:%d, ctx->frame_size:%d",
inputSize, _audioCodecCtx->channels, _audioCodecCtx->frame_size);
#endif
// older ffmpeg versions didn't accept a const input..
AVPacket pkt;
int got_frm = 0;
av_init_packet(&pkt);
pkt.data = const_cast<uint8_t*>(input);
pkt.size = inputSize;
std::unique_ptr<AVFrame, FrameDeleter> frm(FRAMEALLOC(), FrameDeleter());
if (!frm.get()) {
log_error(_("failed to allocate frame."));
return nullptr;
}
int tmp = avcodec_decode_audio4(_audioCodecCtx, frm.get(), &got_frm, &pkt);
#ifdef GNASH_DEBUG_AUDIO_DECODING
const char* fmtname = av_get_sample_fmt_name(_audioCodecCtx->sample_fmt);
log_debug(" decodeFrame | frm->nb_samples: %d | &got_frm: %d | "
"returned %d | inputSize: %d",
frm->nb_samples, got_frm, tmp, inputSize);
#endif
int plane_size;
if (tmp >= 0 && got_frm) {
int data_size = av_samples_get_buffer_size( &plane_size,
_audioCodecCtx->channels, frm->nb_samples,
_audioCodecCtx->sample_fmt, 1);
if (static_cast<int>(outSize) < data_size) {
log_error(_("output buffer size is too small for the current frame "
"(%d < %d)"), outSize, data_size);
return nullptr;
}
memcpy(outPtr, frm->extended_data[0], plane_size);
#if !(defined(HAVE_SWRESAMPLE_H) || defined(HAVE_AVRESAMPLE_H))
int planar = av_sample_fmt_is_planar(_audioCodecCtx->sample_fmt);
if (planar && _audioCodecCtx->channels > 1) {
uint8_t *out = ((uint8_t *)outPtr) + plane_size;
for (int ch = 1; ch < _audioCodecCtx->channels; ch++) {
memcpy(out, frm->extended_data[ch], plane_size);
out += plane_size;
}
}
#endif
outSize = data_size;
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug(" decodeFrame | fmt: %d | fmt_name: %s | planar: %d | "
"plane_size: %d | outSize: %d",
_audioCodecCtx->sample_fmt, fmtname, planar, plane_size, outSize);
#endif
} else {
if (tmp < 0)
log_error(_("avcodec_decode_audio returned %d."), tmp);
if (outSize < 2)
log_error(_("outputSize:%d after decoding %d bytes of input audio "
"data."), outputSize, inputSize);
log_error(_("Upgrading ffmpeg/libavcodec might fix this issue."));
outputSize = 0;
return nullptr;
}
// Resampling is needed.
if (_resampler.init(_audioCodecCtx)) {
// Resampling is needed.
// Compute new size based on frame_size and
// resampling configuration
double resampleFactor = (44100.0/_audioCodecCtx->sample_rate) * (2.0/_audioCodecCtx->channels);
bool stereo = _audioCodecCtx->channels > 1 ? true : false;
int inSamples = stereo ? outSize >> 2 : outSize >> 1;
int expectedMaxOutSamples = std::ceil(inSamples*resampleFactor);
// *channels *sampleSize
int resampledFrameSize = expectedMaxOutSamples*2*2;
// Allocate just the required amount of bytes
std::uint8_t* resampledOutput = new std::uint8_t[resampledFrameSize];
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug(" decodeFrame | Calling the resampler, resampleFactor: %d | "
"in %d hz %d ch %d bytes %d samples, %s fmt", resampleFactor,
_audioCodecCtx->sample_rate, _audioCodecCtx->channels, outSize,
inSamples, fmtname);
log_debug(" decodeFrame | out 44100 hz 2 ch %d bytes",
resampledFrameSize);
#endif
int outSamples = _resampler.resample(frm->extended_data, // input
plane_size, // input
frm->nb_samples, // input
&resampledOutput); // output
// make sure to set outPtr *after* we use it as input to the resampler
outPtr = reinterpret_cast<std::int16_t*>(resampledOutput);
#ifdef GNASH_DEBUG_AUDIO_DECODING
log_debug("resampler returned %d samples ", outSamples);
#endif
if (expectedMaxOutSamples < outSamples) {
log_error(_(" --- Computation of resampled samples (%d) < then the actual returned samples (%d)"),
expectedMaxOutSamples, outSamples);
log_debug(" input frame size: %d", outSize);
log_debug(" input sample rate: %d", _audioCodecCtx->sample_rate);
log_debug(" input channels: %d", _audioCodecCtx->channels);
log_debug(" input samples: %d", inSamples);
log_debug(" output sample rate (assuming): %d", 44100);
log_debug(" output channels (assuming): %d", 2);
log_debug(" output samples: %d", outSamples);
/// Memory errors...
abort();
}
// Use the actual number of samples returned, multiplied
// to get size in bytes (not two-byte samples) and for
// stereo?
outSize = outSamples * 2 * 2;
}
