Small cleanup.

diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc
index c1448ad905c35b0f20f118fab2a92d8e5f662a32..f97392ef74046301ffb7cdb88e7a0d72001ce910 100644 (file)
--- 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 */

diff --git a/src/lib/ffmpeg_decoder.h b/src/lib/ffmpeg_decoder.h
index 1429c306b7429acb379a63c94b861321a5d59b73..4bb09cf9d3b64986567abd0370b3f8bdb92fa401 100644 (file)
--- a/src/lib/ffmpeg_decoder.h
+++ b/src/lib/ffmpeg_decoder.h
@@ -63,7 +63,7 @@ private:
        void decode_bitmap_subtitle (AVSubtitleRect const * rect, ContentTimePeriod period);
 
        void maybe_add_subtitle ();
-       boost::shared_ptr<AudioBuffers> deinterleave_audio (boost::shared_ptr<FFmpegAudioStream> stream, uint8_t** data, int size);
+       boost::shared_ptr<AudioBuffers> deinterleave_audio (boost::shared_ptr<FFmpegAudioStream> stream) const;
 
        std::list<ContentTimePeriod> image_subtitles_during (ContentTimePeriod, bool starting) const;
        std::list<ContentTimePeriod> text_subtitles_during (ContentTimePeriod, bool starting) const;