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Sorta fix pts on encoded audio packets. Sync is off, but at least it is close

This commit is contained in:
Isaac Connor 2019-07-07 08:57:22 -04:00
parent 0f35d86efb
commit 94cc85aa36
2 changed files with 54 additions and 60 deletions
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4
src/zm_ffmpeg_camera.cpp
View File

@ -466,8 +466,8 @@ int FfmpegCamera::OpenFfmpeg() {
} else {
Debug(1, "decoder %s hwConfig doesn't match our type: %s, pix_fmt %s.",
mVideoCodec->name,
av_hwdevice_get_type_name(config-device_type),
av_get_pix_fmt_name(config->pix_fmt);
av_hwdevice_get_type_name(config->device_type),
av_get_pix_fmt_name(config->pix_fmt)
);
}
} // end foreach hwconfig

110
src/zm_videostore.cpp
View File

@ -1030,13 +1030,15 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
}
zm_dump_frame(out_frame, "Out frame after resample");
#if 0
// out_frame pts is in the input pkt pts... needs to be adjusted before sending to the encoder
if ( out_frame->pts != AV_NOPTS_VALUE ) {
if ( !audio_first_pts ) {
audio_first_pts = out_frame->pts;
Debug(1, "No video_first_pts setting to %" PRId64, audio_first_pts);
Debug(1, "No audio_first_pts setting to %" PRId64, audio_first_pts);
out_frame->pts = 0;
} else {
// out_frame_pts is in codec->timebase, audio_first_pts is in packet timebase.
out_frame->pts = out_frame->pts - audio_first_pts;
zm_dump_frame(out_frame, "Out frame after pts adjustment");
}
@ -1046,55 +1048,55 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
out_frame->pts = audio_next_pts;
}
audio_next_pts = out_frame->pts + out_frame->nb_samples;
#endif
av_init_packet(&opkt);
#if LIBAVCODEC_VERSION_CHECK(57, 64, 0, 64, 0)
if ( (ret = avcodec_send_frame(audio_out_ctx, out_frame)) < 0 ) {
Error("Could not send frame (error '%s')",
av_make_error_string(ret).c_str());
zm_av_packet_unref(&opkt);
if ( !zm_send_frame(audio_out_ctx, out_frame, opkt) ) {
return 0;
}
if ( (ret = avcodec_receive_packet(audio_out_ctx, &opkt)) < 0 ) {
if ( AVERROR(EAGAIN) == ret ) {
// The codec may need more samples than it has, perfectly valid
Debug(2, "Codec not ready to give us a packet");
} else {
Error("Could not recieve packet (error %d = '%s')", ret,
av_make_error_string(ret).c_str());
}
zm_av_packet_unref(&opkt);
return 0;
}
#else
int data_present;
if ( (ret = avcodec_encode_audio2(
audio_out_ctx, &opkt, out_frame, &data_present)) < 0 ) {
Error("Could not encode frame (error '%s')",
av_make_error_string(ret).c_str());
zm_av_packet_unref(&opkt);
return 0;
}
if ( !data_present ) {
Debug(2, "Not ready to out a frame yet.");
zm_av_packet_unref(&opkt);
return 0;
}
#endif
#if 0
// These should be set by encoder. They may not directly relate to ipkt due to buffering in codec.
opkt.duration = av_rescale_q(opkt.duration,
audio_in_stream->time_base,
audio_out_stream->time_base);
opkt.pts = av_rescale_q(opkt.pts,
audio_in_stream->time_base,
audio_out_stream->time_base);
opkt.dts = av_rescale_q(opkt.dts,
audio_in_stream->time_base,
audio_out_stream->time_base);
#endif
dumpPacket(audio_out_stream, &opkt, "raw opkt");
opkt.duration = av_rescale_q(
opkt.duration,
audio_out_ctx->time_base,
audio_out_stream->time_base);
// Scale the PTS of the outgoing packet to be the correct time base
if ( ipkt->pts != AV_NOPTS_VALUE ) {
if ( !audio_first_pts ) {
opkt.pts = 0;
audio_first_pts = ipkt->pts;
Debug(1, "No audio_first_pts");
} else {
opkt.pts = av_rescale_q(
opkt.pts,
audio_out_ctx->time_base,
audio_out_stream->time_base);
opkt.pts -= audio_first_pts;
Debug(2, "audio opkt.pts = %" PRId64 " from ipkt->pts(%" PRId64 ") - first_pts(%" PRId64 ")",
opkt.pts, ipkt->pts, audio_first_pts);
}
} else {
Debug(2, "opkt.pts = undef");
opkt.pts = AV_NOPTS_VALUE;
}
if ( opkt.dts != AV_NOPTS_VALUE ) {
if ( !audio_first_dts ) {
opkt.dts = 0;
audio_first_dts = opkt.dts;
} else {
opkt.dts = av_rescale_q(
opkt.dts,
audio_out_ctx->time_base,
audio_out_stream->time_base);
opkt.dts -= audio_first_dts;
Debug(2, "opkt.dts = %" PRId64 " from ipkt.dts(%" PRId64 ") - first_dts(%" PRId64 ")",
opkt.dts, ipkt->dts, audio_first_dts);
}
audio_last_dts = opkt.dts;
} else {
opkt.dts = AV_NOPTS_VALUE;
}
} else {
Debug(2,"copying");
@ -1128,16 +1130,6 @@ int VideoStore::writeAudioFramePacket(AVPacket *ipkt) {
}
if ( ipkt->dts != AV_NOPTS_VALUE ) {
// So if the in has no dts assigned... still need an out dts... so we use cur_dts?
#if 0
if ( audio_last_dts >= audio_in_stream->cur_dts ) {
Debug(1, "Resetting audio_last_dts from (%d) to cur_dts (%d)", audio_last_dts, audio_in_stream->cur_dts);
opkt.dts = audio_next_dts + av_rescale_q( audio_in_stream->cur_dts, AV_TIME_BASE_Q, audio_out_stream->time_base);
} else {
opkt.dts = audio_next_dts + av_rescale_q( audio_in_stream->cur_dts - audio_last_dts, AV_TIME_BASE_Q, audio_out_stream->time_base);
}
#endif
if ( !audio_first_dts ) {
opkt.dts = 0;
audio_first_dts = ipkt->dts;
@ -1251,15 +1243,17 @@ int VideoStore::resample_audio() {
}
out_frame->nb_samples = frame_size;
// resampling changes the duration because the timebase is 1/samples
// I think we should be dealing in codec timebases not stream
if ( in_frame->pts != AV_NOPTS_VALUE ) {
out_frame->pkt_duration = av_rescale_q(
in_frame->pkt_duration,
audio_in_stream->time_base,
audio_out_stream->time_base);
audio_in_ctx->time_base,
audio_out_ctx->time_base);
out_frame->pts = av_rescale_q(
in_frame->pts,
audio_in_stream->time_base,
audio_out_stream->time_base);
audio_in_ctx->time_base,
audio_out_ctx->time_base);
zm_dump_frame(out_frame, "Out frame after timestamp conversion");
}
#else
#if defined(HAVE_LIBAVRESAMPLE)