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synced 2026-03-02 08:53:52 -08:00
Added code to flush the encoders, and also write the final frame's audio samples
This commit is contained in:
Jonathan Thomas
@@ -338,7 +338,7 @@ void FFmpegWriter::write_queued_frames()
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{
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// Process all audio frames (in a separate thread)
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if (info.has_audio && audio_st && !queued_audio_frames.empty())
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write_audio_packets();
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write_audio_packets(false);
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// Loop through each queued image frame
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while (!queued_video_frames.empty())
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@@ -435,26 +435,35 @@ void FFmpegWriter::WriteFrame(FileReaderBase* reader, int start, int length)
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// Write the file trailer (after all frames are written)
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void FFmpegWriter::WriteTrailer()
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{
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// Write any remaining queued frames to video file
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write_queued_frames();
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// Process final audio frame (if any)
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if (info.has_audio && audio_st)
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write_audio_packets(true);
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// Experimental: Repeat last frame many times, to pad
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// the end of the video, to ensure the codec does not
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// ignore the final frames.
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if (last_frame)
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{
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// Flush remaining packets
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//av_write_frame(oc, NULL);
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av_interleaved_write_frame(oc, NULL);
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// if (last_frame)
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// {
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// // Create black frame
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// tr1::shared_ptr<Frame> padding_frame(new Frame(999999, last_frame->GetWidth(), last_frame->GetHeight(), "#000000", last_frame->GetAudioSamplesCount(), last_frame->GetAudioChannelsCount()));
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// padding_frame->AddColor(last_frame->GetWidth(), last_frame->GetHeight(), "#000000");
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//
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// // Add the black frame many times
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// for (int p = 0; p < 100; p++)
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// WriteFrame(padding_frame);
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//
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// // Write these blank frames
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// write_queued_frames();
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// }
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// Create black frame
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//tr1::shared_ptr<Frame> padding_frame(new Frame(999999, last_frame->GetWidth(), last_frame->GetHeight(), "#000000", last_frame->GetAudioSamplesCount(), last_frame->GetAudioChannelsCount()));
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//padding_frame->AddColor(last_frame->GetWidth(), last_frame->GetHeight(), "#000000");
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// Flush encoders (who sometimes hold on to frames)
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flush_encoders();
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// Add the black frame many times
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//for (int p = 0; p < 25; p++)
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// WriteFrame(padding_frame);
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}
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// Write any remaining queued frames to video file
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write_queued_frames();
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// Flush packets
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//av_interleaved_write_frame(oc, NULL);
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/* write the trailer, if any. The trailer must be written
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* before you close the CodecContexts open when you wrote the
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@@ -463,6 +472,120 @@ void FFmpegWriter::WriteTrailer()
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av_write_trailer(oc);
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}
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// Flush encoders
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void FFmpegWriter::flush_encoders()
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{
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if (info.has_audio && audio_codec && audio_st->codec->codec_type == AVMEDIA_TYPE_AUDIO && audio_codec->frame_size <= 1)
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return;
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if (info.has_video && video_st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (oc->oformat->flags & AVFMT_RAWPICTURE) && video_codec->codec->id == CODEC_ID_RAWVIDEO)
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return;
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int stop_encoding = 1;
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// FLUSH VIDEO ENCODER
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if (info.has_video)
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for (;;) {
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cout << "Flushing VIDEO buffer!" << endl;
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// Increment PTS (in frames and scaled to the codec's timebase)
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write_video_count += av_rescale_q(1, AVRational{info.fps.den, info.fps.num}, video_codec->time_base);
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AVPacket pkt;
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av_init_packet(&pkt);
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pkt.data = NULL;
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pkt.size = 0;
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/* encode the image */
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int got_packet = 0;
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int error_code = avcodec_encode_video2(video_codec, &pkt, NULL, &got_packet);
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if (error_code < 0) {
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string error_description = av_err2str(error_code);
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cout << "error: " << error_code << ": " << error_description << endl;
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throw ErrorEncodingVideo("Error while flushing video frame", -1);
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}
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if (!got_packet) {
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stop_encoding = 1;
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break;
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}
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// Override PTS (in frames and scaled to the codec's timebase)
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//pkt.pts = write_video_count;
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// set the timestamp
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if (pkt.pts != AV_NOPTS_VALUE)
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pkt.pts = av_rescale_q(pkt.pts, video_codec->time_base, video_st->time_base);
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if (pkt.dts != AV_NOPTS_VALUE)
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pkt.dts = av_rescale_q(pkt.dts, video_codec->time_base, video_st->time_base);
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if (pkt.duration > 0)
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pkt.duration = av_rescale_q(pkt.duration, video_codec->time_base, video_st->time_base);
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pkt.stream_index = video_st->index;
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// Write packet
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int averror = av_interleaved_write_frame(oc, &pkt);
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if (averror != 0) {
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string error_description = av_err2str(averror);
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cout << "error: " << averror << ": " << error_description << endl;
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throw ErrorEncodingVideo("Error while writing video packet to flush encoder", -1);
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}
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}
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// FLUSH AUDIO ENCODER
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if (info.has_audio)
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for (;;) {
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cout << "Flushing AUDIO buffer!" << endl;
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// Increment PTS (in samples and scaled to the codec's timebase)
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write_audio_count += av_rescale_q(audio_codec->frame_size / av_get_bytes_per_sample(audio_codec->sample_fmt), AVRational{1, info.sample_rate}, audio_codec->time_base);
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AVPacket pkt;
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av_init_packet(&pkt);
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pkt.data = NULL;
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pkt.size = 0;
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pkt.pts = pkt.dts = write_audio_count;
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/* encode the image */
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int got_packet = 0;
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int error_code = avcodec_encode_audio2(audio_codec, &pkt, NULL, &got_packet);
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if (error_code < 0) {
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string error_description = av_err2str(error_code);
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cout << "error: " << error_code << ": " << error_description << endl;
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throw ErrorEncodingAudio("Error while flushing audio frame", -1);
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}
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if (!got_packet) {
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stop_encoding = 1;
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break;
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}
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// Since the PTS can change during encoding, set the value again. This seems like a huge hack,
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// but it fixes lots of PTS related issues when I do this.
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pkt.pts = pkt.dts = write_audio_count;
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// Scale the PTS to the audio stream timebase (which is sometimes different than the codec's timebase)
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if (pkt.pts != AV_NOPTS_VALUE)
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pkt.pts = av_rescale_q(pkt.pts, audio_codec->time_base, audio_st->time_base);
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if (pkt.dts != AV_NOPTS_VALUE)
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pkt.dts = av_rescale_q(pkt.dts, audio_codec->time_base, audio_st->time_base);
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if (pkt.duration > 0)
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pkt.duration = av_rescale_q(pkt.duration, audio_codec->time_base, audio_st->time_base);
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// set stream
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pkt.stream_index = audio_st->index;
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pkt.flags |= AV_PKT_FLAG_KEY;
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// Write packet
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int averror = av_write_frame(oc, &pkt);
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if (averror != 0) {
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string error_description = av_err2str(averror);
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cout << "error: " << averror << ": " << error_description << endl;
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throw ErrorEncodingAudio("Error while writing audio packet to flush encoder", -1);
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}
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}
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}
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// Close the video codec
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void FFmpegWriter::close_video(AVFormatContext *oc, AVStream *st)
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{
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@@ -746,7 +869,7 @@ void FFmpegWriter::open_video(AVFormatContext *oc, AVStream *st)
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}
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// write all queued frames' audio to the video file
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void FFmpegWriter::write_audio_packets()
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void FFmpegWriter::write_audio_packets(bool final)
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{
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// Create a resampler (only once)
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if (!resampler)
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@@ -804,10 +927,9 @@ void FFmpegWriter::write_audio_packets()
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int remaining_frame_samples = total_frame_samples;
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int samples_position = 0;
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// Re-sample audio samples (into additional channels or changing the sample format / number format)
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// The sample rate has already been resampled using the GetInterleavedAudioSamples method.
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if (audio_codec->sample_fmt != AV_SAMPLE_FMT_S16 || info.channels != channels_in_frame) {
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if (!final && (audio_codec->sample_fmt != AV_SAMPLE_FMT_S16 || info.channels != channels_in_frame)) {
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// Audio needs to be converted
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// Create an audio resample context object (used to convert audio samples)
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@@ -838,7 +960,7 @@ void FFmpegWriter::write_audio_packets()
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}
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// Loop until no more samples
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while (remaining_frame_samples > 0) {
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while (remaining_frame_samples > 0 || final) {
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// Get remaining samples needed for this packet
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int remaining_packet_samples = audio_input_frame_size - audio_input_position;
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@@ -850,7 +972,8 @@ void FFmpegWriter::write_audio_packets()
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diff = remaining_frame_samples;
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// Copy frame samples into the packet samples array
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memcpy(samples + audio_input_position, frame_samples + samples_position, diff * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
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if (!final)
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memcpy(samples + audio_input_position, frame_samples + samples_position, diff * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
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// Increment counters
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audio_input_position += diff;
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@@ -859,20 +982,19 @@ void FFmpegWriter::write_audio_packets()
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remaining_packet_samples -= diff;
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// Do we have enough samples to proceed?
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if (audio_input_position < audio_input_frame_size)
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if (audio_input_position < audio_input_frame_size && !final)
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// Not enough samples to encode... so wait until the next frame
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break;
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// Increment PTS (in samples and scaled to the codec's timebase)
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write_audio_count += av_rescale_q(audio_codec->frame_size, AVRational{1, info.sample_rate}, audio_codec->time_base);
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write_audio_count += av_rescale_q(audio_input_position / av_get_bytes_per_sample(audio_codec->sample_fmt), AVRational{1, info.sample_rate}, audio_codec->time_base);
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// Create AVFrame (and fill it with samples)
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AVFrame *frame_final = avcodec_alloc_frame();
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frame_final->nb_samples = audio_codec->frame_size;
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frame_final->nb_samples = audio_input_position / av_get_bytes_per_sample(audio_codec->sample_fmt);
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frame_final->pts = write_audio_count; // Set the AVFrame's PTS
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avcodec_fill_audio_frame(frame_final, audio_codec->channels, audio_codec->sample_fmt, (uint8_t *) samples,
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audio_input_position * av_get_bytes_per_sample(audio_codec->sample_fmt) * audio_codec->channels, 0);
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audio_input_position * av_get_bytes_per_sample(audio_codec->sample_fmt), 0);
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// Init the packet
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AVPacket pkt;
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@@ -928,6 +1050,7 @@ void FFmpegWriter::write_audio_packets()
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// Reset position
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audio_input_position = 0;
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final = false;
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}
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// Delete arrays
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@@ -1084,6 +1207,10 @@ void FFmpegWriter::write_video_packet(tr1::shared_ptr<Frame> frame, AVFrame* fra
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/* if zero size, it means the image was buffered */
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if (error_code == 0 && got_packet_ptr) {
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// Since the PTS can change during encoding, set the value again. This seems like a huge hack,
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// but it fixes lots of PTS related issues when I do this.
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//pkt.pts = pkt.dts = write_video_count;
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// set the timestamp
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if (pkt.pts != AV_NOPTS_VALUE)
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pkt.pts = av_rescale_q(pkt.pts, video_codec->time_base, video_st->time_base);
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