diff options
| author | Carl Hetherington <cth@carlh.net> | 2016-02-09 00:25:20 +0000 |
|---|---|---|
| committer | Carl Hetherington <cth@carlh.net> | 2016-02-09 00:25:20 +0000 |
| commit | 1c996bf18ee3731e3533c82dae98e5abaa336a32 (patch) | |
| tree | 7d38e646cc3a9c668dee16b8af55dbaeeee5451f /src/lib/ffmpeg_decoder.cc | |
| parent | 0b520c4aaca4acb9cdf6334858b462d1a657209d (diff) | |
Small cleanup.
Diffstat (limited to 'src/lib/ffmpeg_decoder.cc')
| -rw-r--r-- | src/lib/ffmpeg_decoder.cc | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc index c1448ad90..f97392ef7 100644 --- a/src/lib/ffmpeg_decoder.cc +++ b/src/lib/ffmpeg_decoder.cc @@ -129,10 +129,14 @@ FFmpegDecoder::pass (PassReason reason, bool accurate) * Only the first buffer will be used for non-planar data, otherwise there will be one per channel. */ shared_ptr<AudioBuffers> -FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t** data, int size) +FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream) const { DCPOMATIC_ASSERT (bytes_per_audio_sample (stream)); + int const size = av_samples_get_buffer_size ( + 0, stream->stream(_format_context)->codec->channels, _frame->nb_samples, audio_sample_format (stream), 1 + ); + /* Deinterleave and convert to float */ /* total_samples and frames will be rounded down here, so if there are stray samples at the end @@ -145,7 +149,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t switch (audio_sample_format (stream)) { case AV_SAMPLE_FMT_U8: { - uint8_t* p = reinterpret_cast<uint8_t *> (data[0]); + uint8_t* p = reinterpret_cast<uint8_t *> (_frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -162,7 +166,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t case AV_SAMPLE_FMT_S16: { - int16_t* p = reinterpret_cast<int16_t *> (data[0]); + int16_t* p = reinterpret_cast<int16_t *> (_frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -179,7 +183,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t case AV_SAMPLE_FMT_S16P: { - int16_t** p = reinterpret_cast<int16_t **> (data); + int16_t** p = reinterpret_cast<int16_t **> (_frame->data); for (int i = 0; i < stream->channels(); ++i) { for (int j = 0; j < frames; ++j) { audio->data(i)[j] = static_cast<float>(p[i][j]) / (1 << 15); @@ -190,7 +194,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t case AV_SAMPLE_FMT_S32: { - int32_t* p = reinterpret_cast<int32_t *> (data[0]); + int32_t* p = reinterpret_cast<int32_t *> (_frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -207,7 +211,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t case AV_SAMPLE_FMT_FLT: { - float* p = reinterpret_cast<float*> (data[0]); + float* p = reinterpret_cast<float*> (_frame->data[0]); int sample = 0; int channel = 0; for (int i = 0; i < total_samples; ++i) { @@ -224,7 +228,7 @@ FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream, uint8_t case AV_SAMPLE_FMT_FLTP: { - float** p = reinterpret_cast<float**> (data); + float** p = reinterpret_cast<float**> (_frame->data); /* Sometimes there aren't as many channels in the _frame as in the stream */ for (int i = 0; i < _frame->channels; ++i) { memcpy (audio->data(i), p[i], frames * sizeof(float)); @@ -334,11 +338,7 @@ FFmpegDecoder::decode_audio_packet () av_q2d ((*stream)->stream (_format_context)->time_base)) + _pts_offset; - int const data_size = av_samples_get_buffer_size ( - 0, (*stream)->stream(_format_context)->codec->channels, _frame->nb_samples, audio_sample_format (*stream), 1 - ); - - shared_ptr<AudioBuffers> data = deinterleave_audio (*stream, _frame->data, data_size); + shared_ptr<AudioBuffers> data = deinterleave_audio (*stream); if (ct < ContentTime ()) { /* Discard audio data that comes before time 0 */ |