X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=ffmpeg_capture.cpp;h=235393a5058b0a2215a8ea021becc6c1fc75ab9a;hb=a5746714e6ca1e665bf9e74344e67712443f947a;hp=c79ecf337fe8213efe16b26f3e3cd436c5c4dfe5;hpb=ed097a5749ba8b78ffc7886aa1362c0095234ab6;p=nageru diff --git a/ffmpeg_capture.cpp b/ffmpeg_capture.cpp index c79ecf3..235393a 100644 --- a/ffmpeg_capture.cpp +++ b/ffmpeg_capture.cpp @@ -208,7 +208,6 @@ YCbCrFormat decode_ycbcr_format(const AVPixFmtDescriptor *desc, const AVFrame *f FFmpegCapture::FFmpegCapture(const string &filename, unsigned width, unsigned height) : filename(filename), width(width), height(height), video_timebase{1, 1} { - // Not really used for anything. description = "Video: " + filename; last_frame = steady_clock::now(); @@ -283,9 +282,15 @@ void FFmpegCapture::producer_thread_func() pthread_setname_np(pthread_self(), thread_name); while (!producer_thread_should_quit.should_quit()) { - string pathname = search_for_file(filename); - if (filename.empty()) { - fprintf(stderr, "%s not found, sleeping one second and trying again...\n", filename.c_str()); + string filename_copy; + { + lock_guard lock(filename_mu); + filename_copy = filename; + } + + string pathname = search_for_file(filename_copy); + if (pathname.empty()) { + fprintf(stderr, "%s not found, sleeping one second and trying again...\n", filename_copy.c_str()); send_disconnected_frame(); producer_thread_should_quit.sleep_for(seconds(1)); continue; @@ -322,6 +327,7 @@ void FFmpegCapture::send_disconnected_frame() if (pixel_format == bmusb::PixelFormat_8BitBGRA) { video_format.stride = width * 4; video_frame.len = width * height * 4; + memset(video_frame.data, 0, video_frame.len); } else { video_format.stride = width; current_frame_ycbcr_format.luma_coefficients = YCBCR_REC_709; @@ -334,8 +340,9 @@ void FFmpegCapture::send_disconnected_frame() current_frame_ycbcr_format.cr_x_position = 0.0f; current_frame_ycbcr_format.cr_y_position = 0.0f; video_frame.len = width * height * 2; + memset(video_frame.data, 0, width * height); + memset(video_frame.data + width * height, 128, width * height); // Valid for both NV12 and planar. } - memset(video_frame.data, 0, video_frame.len); frame_callback(-1, AVRational{1, TIMEBASE}, -1, AVRational{1, TIMEBASE}, timecode++, video_frame, /*video_offset=*/0, video_format, @@ -491,6 +498,38 @@ bool FFmpegCapture::play_video(const string &pathname) int64_t audio_pts_as_video_pts = av_rescale_q(audio_pts, audio_timebase, video_timebase); audio_frame->received_timestamp = compute_frame_start(audio_pts_as_video_pts, pts_origin, video_timebase, start, rate); + if (audio_frame->len != 0) { + // The received timestamps in Nageru are measured after we've just received the frame. + // However, pts (especially audio pts) is at the _beginning_ of the frame. + // If we have locked audio, the distinction doesn't really matter, as pts is + // on a relative scale and a fixed offset is fine. But if we don't, we will have + // a different number of samples each time, which will cause huge audio jitter + // and throw off the resampler. + // + // In a sense, we should have compensated by adding the frame and audio lengths + // to video_frame->received_timestamp and audio_frame->received_timestamp respectively, + // but that would mean extra waiting in sleep_until(). All we need is that they + // are correct relative to each other, though (and to the other frames we send), + // so just align the end of the audio frame, and we're fine. + size_t num_samples = (audio_frame->len * 8) / audio_format.bits_per_sample / audio_format.num_channels; + double offset = double(num_samples) / OUTPUT_FREQUENCY - + double(video_format.frame_rate_den) / video_format.frame_rate_nom; + audio_frame->received_timestamp += duration_cast(duration(offset)); + } + + steady_clock::time_point now = steady_clock::now(); + if (duration(now - next_frame_start).count() >= 0.1) { + // If we don't have enough CPU to keep up, or if we have a live stream + // where the initial origin was somehow wrong, we could be behind indefinitely. + // In particular, this will give the audio resampler problems as it tries + // to speed up to reduce the delay, hitting the low end of the buffer every time. + fprintf(stderr, "%s: Playback %.0f ms behind, resetting time scale\n", + pathname.c_str(), + 1e3 * duration(now - next_frame_start).count()); + pts_origin = frame->pts; + start = next_frame_start = now; + timecode += MAX_FPS * 2 + 1; + } bool finished_wakeup = producer_thread_should_quit.sleep_until(next_frame_start); if (finished_wakeup) { if (audio_frame->len > 0) { @@ -597,6 +636,7 @@ AVFrameWithDeleter FFmpegCapture::decode_frame(AVFormatContext *format_ctx, AVCo AVFrameWithDeleter video_avframe = av_frame_alloc_unique(); bool eof = false; *audio_pts = -1; + bool has_audio = false; do { AVPacket pkt; unique_ptr pkt_cleanup( @@ -615,9 +655,7 @@ AVFrameWithDeleter FFmpegCapture::decode_frame(AVFormatContext *format_ctx, AVCo return AVFrameWithDeleter(nullptr); } } else if (pkt.stream_index == audio_stream_index) { - if (*audio_pts == -1) { - *audio_pts = pkt.pts; - } + has_audio = true; if (avcodec_send_packet(audio_codec_ctx, &pkt) < 0) { fprintf(stderr, "%s: Cannot send packet to audio codec.\n", pathname.c_str()); *error = true; @@ -629,10 +667,13 @@ AVFrameWithDeleter FFmpegCapture::decode_frame(AVFormatContext *format_ctx, AVCo } // Decode audio, if any. - if (*audio_pts != -1) { + if (has_audio) { for ( ;; ) { int err = avcodec_receive_frame(audio_codec_ctx, audio_avframe.get()); if (err == 0) { + if (*audio_pts == -1) { + *audio_pts = audio_avframe->pts; + } convert_audio(audio_avframe.get(), audio_frame, audio_format); } else if (err == AVERROR(EAGAIN)) { break;