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ffmpeg: Simplify ffmpegAudioCursor.cxx a little

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This refactors the `reload_buffer` loop so all of the decode
logic is inside the loop, and all of the resampling logic is
outside.
This commit is contained in:
Sam Edwards 2018-03-31 02:54:48 -06:00
parent eb591674e0
commit 4ffd364415
1 changed files with 54 additions and 58 deletions
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112
panda/src/ffmpeg/ffmpegAudioCursor.cxx
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@ -297,71 +297,67 @@ fetch_packet() {
*/ */
bool FfmpegAudioCursor:: bool FfmpegAudioCursor::
reload_buffer() { reload_buffer() {
int got_frame = 0;
while (_buffer_head == _buffer_tail) { while (!got_frame) {
// If we're out of packets, generate silence. // If we're out of packets, generate silence.
if (_packet->data == 0) { if (_packet->data == nullptr) {
_buffer_head = 0; _buffer_head = 0;
_buffer_tail = _buffer_size; _buffer_tail = _buffer_size;
memset(_buffer, 0, _buffer_size * 2); memset(_buffer, 0, _buffer_size * 2);
return true; return true;
} else if (_packet_size > 0) { } else if (_packet_size == 0) {
int bufsize = _buffer_size * 2;
int got_frame;
AVPacket *pkt;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 12, 100)
pkt = av_packet_alloc();
#else
AVPacket _pkt;
pkt = &_pkt;
av_init_packet(pkt);
#endif
pkt->data = _packet_data;
pkt->size = _packet_size;
int len = avcodec_decode_audio4(_audio_ctx, _frame, &got_frame, pkt);
movies_debug("avcodec_decode_audio4 returned " << len);
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 12, 100)
av_packet_free(&pkt);
#else
av_free_packet(pkt);
#endif
bufsize = 0;
if (got_frame) {
#ifdef HAVE_SWRESAMPLE
if (_resample_ctx) {
// Resample the data to signed 16-bit sample format.
bufsize = swr_convert(_resample_ctx, (uint8_t **)&_buffer, _buffer_size / 2, (const uint8_t**)_frame->extended_data, _frame->nb_samples);
bufsize *= _audio_channels * 2;
} else
#endif
{
bufsize = _frame->linesize[0];
memcpy(_buffer, _frame->data[0], bufsize);
}
}
#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52, 19, 100)
av_frame_unref(_frame);
#endif
if (len < 0) {
return false;
} else if (len == 0){
return true;
}
_packet_data += len;
_packet_size -= len;
if (bufsize > 0) {
_buffer_head = 0;
_buffer_tail = (bufsize/2);
return true;
}
} else {
fetch_packet(); fetch_packet();
} }
AVPacket *pkt;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 12, 100)
pkt = av_packet_alloc();
#else
AVPacket _pkt;
pkt = &_pkt;
av_init_packet(pkt);
#endif
pkt->data = _packet_data;
pkt->size = _packet_size;
int len = avcodec_decode_audio4(_audio_ctx, _frame, &got_frame, pkt);
movies_debug("avcodec_decode_audio4 returned " << len);
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 12, 100)
av_packet_free(&pkt);
#else
av_free_packet(pkt);
#endif
if (len < 0) {
return false;
} else if (len == 0) {
return true;
}
_packet_data += len;
_packet_size -= len;
}
int bufsize;
#ifdef HAVE_SWRESAMPLE
if (_resample_ctx) {
// Resample the data to signed 16-bit sample format.
bufsize = swr_convert(_resample_ctx, (uint8_t **)&_buffer, _buffer_size / 2, (const uint8_t**)_frame->extended_data, _frame->nb_samples);
bufsize *= _audio_channels * 2;
} else
#endif
{
bufsize = _frame->linesize[0];
memcpy(_buffer, _frame->data[0], bufsize);
}
#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52, 19, 100)
av_frame_unref(_frame);
#endif
if (bufsize > 0) {
_buffer_head = 0;
_buffer_tail = (bufsize/2);
return true;
} }
return true; return true;
} }