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just more messing around trying to get audio transcoding to work

pull/1185/head
Isaac Connor 2016-09-26 17:04:04 -04:00
parent 06da04c5d7
commit 1bd652ce4b
3 changed files with 94 additions and 54 deletions
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25
src/zm_ffmpeg_camera.cpp
View File

@ -85,6 +85,8 @@ FfmpegCamera::FfmpegCamera( int p_id, const std::string &p_path, const std::stri
FfmpegCamera::~FfmpegCamera()
{
CloseFfmpeg();
if ( capture )
@ -575,15 +577,20 @@ Debug(5, "After av_read_frame (%d)", ret );
Error( "Unable to read packet from stream %d: error %d \"%s\".", packet.stream_index, ret, errbuf );
return( -1 );
}
Debug( 3, "Got packet from stream %d dts (%d) pts(%d) key?(%d)", packet.stream_index, packet.dts, packet.pts, packet.flags & AV_PKT_FLAG_KEY );
//av_packet_ref( &packet, &packet );
int key_frame = packet.flags & AV_PKT_FLAG_KEY;
Debug( 3, "Got packet from stream %d packet pts (%d) dts(%d), key?(%d)",
packet.stream_index, packet.pts, packet.dts,
key_frame
);
//Video recording
if ( recording ) {
// The directory we are recording to is no longer tied to the current event.
// Need to re-init the videostore with the correct directory and start recording again
// for efficiency's sake, we should test for keyframe before we test for directory change...
if ( videoStore && (packet.flags & AV_PKT_FLAG_KEY) && (strcmp(oldDirectory, event_file) != 0 ) ) {
if ( videoStore && key_frame && (strcmp(oldDirectory, event_file) != 0 ) ) {
// don't open new videostore until we're on a key frame..would this require an offset adjustment for the event as a result?...
// if we store our key frame location with the event will that be enough?
Info("Re-starting video storage module");
@ -638,7 +645,7 @@ Debug(5, "After av_read_frame (%d)", ret );
while ( ( queued_packet = packetqueue.popPacket() ) ) {
packet_count += 1;
//Write the packet to our video store
Debug(2, "Writing queued packet stream: %d KEY %d, remaining (%d)", queued_packet->stream_index, queued_packet->flags & AV_PKT_FLAG_KEY, packetqueue.size() );
Debug(2, "Writing queued packet stream: %d KEY %d, remaining (%d)", queued_packet->stream_index, queued_packet->flags & AV_PKT_FLAG_KEY, packetqueue.size() );
if ( queued_packet->stream_index == mVideoStreamId ) {
ret = videoStore->writeVideoFramePacket( queued_packet );
} else if ( queued_packet->stream_index == mAudioStreamId ) {
@ -666,17 +673,21 @@ Debug(5, "After av_read_frame (%d)", ret );
//Buffer video packets, since we are not recording. All audio packets are keyframes, so only if it's a video keyframe
if ( packet.stream_index == mVideoStreamId) {
if ( packet.flags & AV_PKT_FLAG_KEY ) {
if ( key_frame ) {
Debug(3, "Clearing queue");
packetqueue.clearQueue();
}
if ( packet.pts && video_last_pts > packet.pts ) {
Debug(3, "Clearing queue due to out of order pts");
Warning( "Clearing queue due to out of order pts");
packetqueue.clearQueue();
}
}
if ( packet.stream_index != mAudioStreamId || record_audio ) {
if (
( packet.stream_index != mAudioStreamId || record_audio )
&&
( key_frame || packetqueue.size() )
) {
packetqueue.queuePacket( &packet );
}
} // end if recording or not

119
src/zm_videostore.cpp
View File

@ -52,7 +52,6 @@ VideoStore::VideoStore(const char *filename_in, const char *format_in,
Info("Opening video storage stream %s format: %s\n", filename, format);
int ret;
static char error_buffer[255];
//Init everything we need, shouldn't have to do this, ffmpeg_camera or something else will call it.
//av_register_all();
@ -116,6 +115,10 @@ VideoStore::VideoStore(const char *filename_in, const char *format_in,
}
#endif
// Just copy them from the input, no reason to choose different
video_output_context->time_base = video_input_context->time_base;
video_output_stream->time_base = video_input_stream->time_base;
Debug(3, "Time bases: VIDEO input stream (%d/%d) input codec: (%d/%d) output stream: (%d/%d) output codec (%d/%d)",
video_input_stream->time_base.num,
video_input_stream->time_base.den,
@ -260,17 +263,8 @@ Debug(2, "Have audio_output_context");
audio_output_context->sample_fmt = AV_SAMPLE_FMT_FLTP;
}
Debug(3, "Audio Time bases input stream (%d/%d) input codec: (%d/%d) output_stream (%d/%d) output codec (%d/%d)",
audio_input_stream->time_base.num,
audio_input_stream->time_base.den,
audio_input_context->time_base.num,
audio_input_context->time_base.den,
audio_output_stream->time_base.num,
audio_output_stream->time_base.den,
audio_output_context->time_base.num,
audio_output_context->time_base.den
);
audio_output_stream->time_base = (AVRational){ 1, audio_output_context->sample_rate };
//audio_output_stream->time_base = audio_input_stream->time_base;
audio_output_context->time_base = (AVRational){ 1, audio_output_context->sample_rate };
Debug(3, "Audio Time bases input stream (%d/%d) input codec: (%d/%d) output_stream (%d/%d) output codec (%d/%d)",
audio_input_stream->time_base.num,
@ -282,9 +276,6 @@ Debug(2, "Have audio_output_context");
audio_output_context->time_base.num,
audio_output_context->time_base.den
);
/** Set the sample rate for the container. */
//audio_output_stream->time_base.den = audio_input_context->sample_rate;
//audio_output_stream->time_base.num = 1;
ret = avcodec_open2(audio_output_context, audio_output_codec, &opts );
if ( ret < 0 ) {
@ -409,19 +400,11 @@ Debug(2, "Have audio_output_context");
Debug(3, "Audio is mono");
}
} // end if is AAC
if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
audio_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
Debug(3, "Audio Time bases input stream time base(%d/%d) input codec tb: (%d/%d) video_output_stream->time-base(%d/%d) output codec tb (%d/%d)",
audio_input_stream->time_base.num,
audio_input_stream->time_base.den,
audio_input_context->time_base.num,
audio_input_context->time_base.den,
audio_output_stream->time_base.num,
audio_output_stream->time_base.den,
audio_output_context->time_base.num,
audio_output_context->time_base.den
);
if (oc->oformat->flags & AVFMT_GLOBALHEADER) {
audio_output_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
} else {
Debug(3, "No Audio output stream");
audio_output_stream = NULL;
@ -469,6 +452,40 @@ Debug(2, "Have audio_output_context");
VideoStore::~VideoStore(){
if ( audio_output_codec ) {
Debug(1, "Have audio encoder, need to flush it's output" );
// Do we need to flush the outputs? I have no idea.
AVPacket pkt;
int got_packet;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
int64_t size;
while(1) {
ret = avcodec_encode_audio2( audio_output_context, &pkt, NULL, &got_packet );
if (ret < 0) {
Error("ERror encoding audio while flushing");
break;
}
Debug(1, "Have audio encoder, need to flush it's output" );
size += pkt.size;
if (!got_packet) {
break;
}
Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts, pkt.dts, pkt.duration );
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, audio_output_context->time_base, audio_output_stream->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, audio_output_context->time_base, audio_output_stream->time_base);
if (pkt.duration > 0)
pkt.duration = av_rescale_q(pkt.duration, audio_output_context->time_base, audio_output_stream->time_base);
Debug(2, "writing flushed packet pts(%d) dts(%d) duration(%d)", pkt.pts, pkt.dts, pkt.duration );
pkt.stream_index = audio_output_stream->index;
av_interleaved_write_frame( oc, &pkt );
zm_av_unref_packet( &pkt );
} // while 1
}
/* Write the trailer before close */
if ( int rc = av_write_trailer(oc) ) {
Error("Error writing trailer %s", av_err2str( rc ) );
@ -626,7 +643,6 @@ Debug(4, "Not video and RAWPICTURE");
dumpPacket(&opkt);
} else {
int ret;
prevDts = opkt.dts; // Unsure if av_interleaved_write_frame() clobbers opkt.dts when out of order, so storing in advance
ret = av_interleaved_write_frame(oc, &opkt);
@ -654,15 +670,12 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
return -1;*/
//zm_dump_stream_format( oc, ipkt->stream_index, 0, 1 );
int ret;
av_init_packet(&opkt);
Debug(5, "after init packet" );
#if 1
//Scale the PTS of the outgoing packet to be the correct time base
if (ipkt->pts != AV_NOPTS_VALUE) {
if ( ipkt->pts != AV_NOPTS_VALUE ) {
if ( (!audio_start_pts) || ( audio_start_pts > ipkt->pts ) ) {
Debug(1, "Resetting audeo_start_pts from (%d) to (%d)", audio_start_pts, ipkt->pts );
//never gets set, so the first packet can set it.
@ -677,7 +690,7 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
//Scale the DTS of the outgoing packet to be the correct time base
if(ipkt->dts == AV_NOPTS_VALUE) {
if ( (!audio_start_dts) || (audio_start_dts > audio_input_stream->cur_dts ) ) {
Debug(1, "Resetting audeo_start_pts from (%d) to (%d)", audio_start_dts, audio_input_stream->cur_dts );
Debug(1, "Resetting audio_start_pts from (%d) to cur_dts (%d)", audio_start_dts, audio_input_stream->cur_dts );
audio_start_dts = audio_input_stream->cur_dts;
}
opkt.dts = av_rescale_q(audio_input_stream->cur_dts - audio_start_dts, AV_TIME_BASE_Q, audio_output_stream->time_base);
@ -685,7 +698,7 @@ int VideoStore::writeAudioFramePacket( AVPacket *ipkt ) {
opkt.dts, audio_input_stream->cur_dts, audio_start_dts
);
} else {
if ( (!audio_start_dts) || ( audio_start_dts > ipkt->dts ) ) {
if ( ( ! audio_start_dts ) || ( audio_start_dts > ipkt->dts ) ) {
Debug(1, "Resetting audeo_start_dts from (%d) to (%d)", audio_start_dts, ipkt->dts );
audio_start_dts = ipkt->dts;
}
@ -750,8 +763,6 @@ av_codec_is_encoder( audio_output_context->codec)
av_frame_unref( input_frame );
#else
// convert the packet to the codec timebase from the stream timebase
av_packet_rescale_ts( ipkt, audio_input_stream->time_base, audio_input_context->time_base );
/**
* Decode the audio frame stored in the packet.
@ -802,11 +813,6 @@ av_codec_is_encoder( audio_output_context->codec)
return 0;
}
/** Create a new frame to store the audio samples. */
if (!(output_frame = zm_av_frame_alloc())) {
Error("Could not allocate output frame");
return 0;
}
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
@ -834,15 +840,19 @@ av_codec_is_encoder( audio_output_context->codec)
return 0;
}
/** Set a timestamp based on the sample rate for the container. */
if (output_frame) {
output_frame->pts = av_frame_get_best_effort_timestamp(output_frame);
}
Debug(4, "About to read");
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
Error( "Could not read data from FIFO\n");
return 0;
}
/** Set a timestamp based on the sample rate for the container. */
output_frame->pts = av_rescale_q( opkt.pts, audio_output_context->time_base, audio_output_stream->time_base );
// convert the packet to the codec timebase from the stream timebase
Debug(3, "output_frame->pts(%d) best effort(%d)", output_frame->pts,
av_frame_get_best_effort_timestamp(output_frame)
);
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
@ -859,9 +869,24 @@ av_codec_is_encoder( audio_output_context->codec)
zm_av_unref_packet(&opkt);
return 0;
}
Debug(2, "opkt dts (%d) pts(%d) duration:(%d)", opkt.dts, opkt.pts, opkt.duration );
// Convert tb from code back to stream
av_packet_rescale_ts(&opkt, audio_output_context->time_base, audio_output_stream->time_base);
//av_packet_rescale_ts(&opkt, audio_output_context->time_base, audio_output_stream->time_base);
if (opkt.pts != AV_NOPTS_VALUE) {
opkt.pts = av_rescale_q( opkt.pts, audio_output_context->time_base, audio_output_stream->time_base);
}
if ( opkt.dts != AV_NOPTS_VALUE)
opkt.dts = av_rescale_q( opkt.dts, audio_output_context->time_base, audio_output_stream->time_base);
if ( opkt.duration > 0)
opkt.duration = av_rescale_q( opkt.duration, audio_output_context->time_base, audio_output_stream->time_base);
Debug(2, "opkt dts (%d) pts(%d) duration:(%d) pos(%d) ", opkt.dts, opkt.pts, opkt.duration, opkt.pos );
//opkt.dts = AV_NOPTS_VALUE;
#endif
} else {

4
src/zm_videostore.h
View File

@ -32,6 +32,7 @@ private:
AVCodecContext *video_input_context;
AVCodecContext *audio_input_context;
int ret;
// The following are used when encoding the audio stream to AAC
AVCodec *audio_output_codec;
@ -53,6 +54,9 @@ uint8_t *converted_input_samples = NULL;
int64_t audio_start_pts;
int64_t audio_start_dts;
int64_t start_pts;
int64_t start_dts;
int64_t prevDts;
int64_t filter_in_rescale_delta_last;