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hack to make sync work on resample

This commit is contained in:
Isaac Connor 2019-09-12 17:59:08 -04:00
parent d9d4c6b4ca
commit 788b5b480b
1 changed files with 17 additions and 11 deletions
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22
src/zm_videostore.cpp
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@ -445,6 +445,7 @@ VideoStore::~VideoStore() {
* At the end of the file, we pass the remaining samples to
* the encoder. */
int frame_size = audio_out_ctx->frame_size;
Debug(2, "av_audio_fifo_size = %s", av_audio_fifo_size(fifo));
while ( av_audio_fifo_size(fifo) > 0 ) {
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
@ -454,10 +455,7 @@ VideoStore::~VideoStore() {
// SHould probably set the frame size to what is reported FIXME
if ( av_audio_fifo_read(fifo, (void **)out_frame->data, frame_size) ) {
av_init_packet(&opkt);
if ( zm_send_frame(audio_out_ctx, out_frame, opkt) ) {
if ( zm_receive_packet(audio_out_ctx, pkt) )
if ( zm_send_frame(audio_out_ctx, out_frame, pkt) ) {
pkt.stream_index = audio_out_stream->index;
pkt.duration = av_rescale_q(
@ -466,7 +464,6 @@ VideoStore::~VideoStore() {
audio_out_stream->time_base);
}
} // end if data returned from fifo
} // end while still data in the fifo
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
@ -980,6 +977,7 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
dumpPacket(audio_in_stream, ipkt, "input packet");
if ( !audio_first_pts ) {
#if 0
// Since audio starts after the start of the video, need to set this here.
audio_first_pts = av_rescale_q(
video_first_pts,
@ -992,8 +990,13 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
audio_first_pts -= audio_next_pts;
audio_next_pts = 0;
}
#else
audio_first_pts = ipkt->pts;
audio_next_pts = audio_out_ctx->frame_size;
#endif
}
if ( !audio_first_dts ) {
#if 0
// Since audio starts after the start of the video, need to set this here.
audio_first_dts = av_rescale_q(
video_first_dts,
@ -1001,11 +1004,14 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
audio_in_stream->time_base
);
audio_next_dts = ipkt->dts - audio_first_dts;
if ( ipkt->dts < 0 ) {
audio_first_dts -= ipkt->dts;
ipkt->dts = 0;
if ( audio_next_dts < 0 ) {
audio_first_dts -= audio_next_dts;
audio_next_dts = 0;
}
Debug(2, "Starting audio first_dts will become %" PRId64, audio_first_dts);
#else
audio_first_dts = ipkt->dts;
#endif
}
// Need to adjust pts before feeding to decoder.... should really copy the pkt instead of modifying it