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-rw-r--r--src/recorder/recorder.c140
1 files changed, 67 insertions, 73 deletions
diff --git a/src/recorder/recorder.c b/src/recorder/recorder.c
index 7ace350..2fc62d5 100644
--- a/src/recorder/recorder.c
+++ b/src/recorder/recorder.c
@@ -273,106 +273,97 @@ static int recorder_setup_video_encoder(gsr_recorder *self) {
return GSR_ERROR_OK;
}
-static int recorder_setup_audio(gsr_recorder *self) {
- if(gsr_audio_capture_init(&self->audio_capture, &self->encoder, self->recording_clock, &self->running) != GSR_ERROR_OK) {
+static int recorder_setup_audio_track(gsr_recorder *self, const gsr_merged_audio_inputs *merged_audio_inputs, int audio_stream_index, gsr_audio_track *audio_track) {
+ const bool use_amix = gsr_audio_inputs_should_use_amix(merged_audio_inputs);
+
+ memset(audio_track, 0, sizeof(*audio_track));
+ audio_track->stream_index = audio_stream_index;
+ audio_track->codec_context = create_audio_codec_context(self->settings.fps, self->settings.audio_codec, use_amix, self->settings.audio_bitrate);
+ if(!audio_track->codec_context)
return GSR_ERROR_GENERIC;
+ AVStream *audio_stream = NULL;
+ if(!self->settings.is_replaying) {
+ audio_stream = create_stream(self->av_format_context, audio_track->codec_context);
+ if(!audio_stream)
+ return GSR_ERROR_GENERIC;
+
+ if(gsr_encoder_add_recording_destination(&self->encoder, audio_track->codec_context, self->av_format_context, audio_stream, 0) == (size_t)-1)
+ gsr_log(GSR_LOG_LEVEL_ERROR, "added too many audio sources");
}
-
- int audio_stream_index = GSR_VIDEO_STREAM_INDEX + 1;
- for(size_t audio_track_index = 0; audio_track_index < self->audio_input_tracks->num_items; ++audio_track_index) {
- const gsr_merged_audio_inputs *merged_audio_inputs = &self->audio_input_tracks->items[audio_track_index];
- const bool use_amix = gsr_audio_inputs_should_use_amix(merged_audio_inputs);
- AVCodecContext *audio_codec_context = create_audio_codec_context(self->settings.fps, self->settings.audio_codec, use_amix, self->settings.audio_bitrate);
- if(!audio_codec_context) {
- return GSR_ERROR_GENERIC;
- }
+ snprintf(audio_track->name, sizeof(audio_track->name), "%s", merged_audio_inputs->track_name);
+ if(audio_stream && audio_track->name[0] != '\0' && !self->settings.exclude_metadata)
+ av_dict_set(&audio_stream->metadata, "title", audio_track->name, 0);
- AVStream *audio_stream = NULL;
- if(!self->settings.is_replaying) {
- audio_stream = create_stream(self->av_format_context, audio_codec_context);
- if(gsr_encoder_add_recording_destination(&self->encoder, audio_codec_context, self->av_format_context, audio_stream, 0) == (size_t)-1)
- gsr_log(GSR_LOG_LEVEL_ERROR, "added too many audio sources");
- }
+ if(!open_audio(audio_track->codec_context, self->settings.ffmpeg_audio_opts))
+ return GSR_ERROR_GENERIC;
- if(audio_stream && merged_audio_inputs->track_name[0] != '\0' && !self->settings.exclude_metadata)
- av_dict_set(&audio_stream->metadata, "title", merged_audio_inputs->track_name, 0);
+ if(audio_stream)
+ avcodec_parameters_from_context(audio_stream->codecpar, audio_track->codec_context);
- if(!open_audio(audio_codec_context, self->settings.ffmpeg_audio_opts)) {
- return GSR_ERROR_GENERIC;
+ #if LIBAVCODEC_VERSION_MAJOR < 60
+ const int num_channels = audio_track->codec_context->channels;
+ #else
+ const int num_channels = audio_track->codec_context->ch_layout.nb_channels;
+ #endif
+ AVFilterContext *src_filter_ctx[GSR_MAX_AUDIO_SOURCES_PER_TRACK];
+ if(use_amix) {
+ if(merged_audio_inputs->num_items > GSR_MAX_AUDIO_SOURCES_PER_TRACK) {
+ gsr_log(GSR_LOG_LEVEL_ERROR, "too many audio sources for one audio track, the maximum is %d", GSR_MAX_AUDIO_SOURCES_PER_TRACK);
+ return GSR_ERROR_GENERIC;
}
- if(audio_stream)
- avcodec_parameters_from_context(audio_stream->codecpar, audio_codec_context);
-
- #if LIBAVCODEC_VERSION_MAJOR < 60
- const int num_channels = audio_codec_context->channels;
- #else
- const int num_channels = audio_codec_context->ch_layout.nb_channels;
- #endif
-
- //audio_frame->sample_rate = audio_codec_context->sample_rate;
-
- AVFilterContext *src_filter_ctx[GSR_MAX_AUDIO_SOURCES_PER_TRACK];
- AVFilterGraph *graph = NULL;
- AVFilterContext *sink = NULL;
- if(use_amix) {
- if(merged_audio_inputs->num_items > GSR_MAX_AUDIO_SOURCES_PER_TRACK) {
- gsr_log(GSR_LOG_LEVEL_ERROR, "too many audio sources for one audio track, the maximum is %d", GSR_MAX_AUDIO_SOURCES_PER_TRACK);
- return GSR_ERROR_GENERIC;
- }
- if(gsr_audio_init_filter_graph(audio_codec_context, &graph, &sink, src_filter_ctx, merged_audio_inputs->num_items) < 0) {
- gsr_log(GSR_LOG_LEVEL_ERROR, "failed to create audio filter");
- return GSR_ERROR_GENERIC;
- }
+ if(gsr_audio_init_filter_graph(audio_track->codec_context, &audio_track->graph, &audio_track->sink, src_filter_ctx, merged_audio_inputs->num_items) < 0) {
+ gsr_log(GSR_LOG_LEVEL_ERROR, "failed to create audio filter");
+ return GSR_ERROR_GENERIC;
}
+ }
- // TODO: Cleanup above
-
- const double audio_fps = (double)audio_codec_context->sample_rate / (double)audio_codec_context->frame_size;
- const double timeout_sec = 1000.0 / audio_fps / 1000.0;
+ const double audio_fps = (double)audio_track->codec_context->sample_rate / (double)audio_track->codec_context->frame_size;
+ const double timeout_sec = 1000.0 / audio_fps / 1000.0;
- const double audio_startup_time_seconds = self->force_no_audio_offset ? 0 : audio_codec_get_desired_delay(self->settings.audio_codec, self->settings.fps);// * ((double)audio_codec_context->frame_size / 1024.0);
- const double num_audio_frames_shift = audio_startup_time_seconds / timeout_sec;
+ const double audio_startup_time_seconds = self->force_no_audio_offset ? 0 : audio_codec_get_desired_delay(self->settings.audio_codec, self->settings.fps);
+ const double num_audio_frames_shift = audio_startup_time_seconds / timeout_sec;
+ audio_track->pts = -audio_track->codec_context->frame_size * num_audio_frames_shift;
- gsr_audio_track audio_track;
- memset(&audio_track, 0, sizeof(audio_track));
- snprintf(audio_track.name, sizeof(audio_track.name), "%s", merged_audio_inputs->track_name);
- audio_track.codec_context = audio_codec_context;
- audio_track.graph = graph;
- audio_track.sink = sink;
- audio_track.stream_index = audio_stream_index;
- audio_track.pts = -audio_codec_context->frame_size * num_audio_frames_shift;
-
- int audio_track_result = GSR_ERROR_OK;
- if(gsr_audio_inputs_has_app_audio(merged_audio_inputs)) {
- assert(!use_amix);
+ if(gsr_audio_inputs_has_app_audio(merged_audio_inputs)) {
+ assert(!use_amix);
#ifdef GSR_APP_AUDIO
- audio_track_result = gsr_audio_track_init_application_input(&audio_track, merged_audio_inputs, audio_codec_context, num_channels, num_audio_frames_shift, self->pipewire_audio);
+ return gsr_audio_track_init_application_input(audio_track, merged_audio_inputs, audio_track->codec_context, num_channels, num_audio_frames_shift, self->pipewire_audio);
+#else
+ return GSR_ERROR_UNSUPPORTED;
#endif
- } else {
- audio_track_result = gsr_audio_track_init_device_inputs(&audio_track, merged_audio_inputs, audio_codec_context, num_channels, num_audio_frames_shift, src_filter_ctx, use_amix);
- }
+ }
+
+ return gsr_audio_track_init_device_inputs(audio_track, merged_audio_inputs, audio_track->codec_context, num_channels, num_audio_frames_shift, src_filter_ctx, use_amix);
+}
+
+static int recorder_setup_audio(gsr_recorder *self) {
+ if(gsr_audio_capture_init(&self->audio_capture, &self->encoder, self->recording_clock, &self->running) != GSR_ERROR_OK)
+ return GSR_ERROR_GENERIC;
+ int audio_stream_index = GSR_VIDEO_STREAM_INDEX + 1;
+ for(size_t i = 0; i < self->audio_input_tracks->num_items; ++i) {
+ gsr_audio_track audio_track;
+ const int audio_track_result = recorder_setup_audio_track(self, &self->audio_input_tracks->items[i], audio_stream_index, &audio_track);
if(audio_track_result != GSR_ERROR_OK) {
+ gsr_audio_track_deinit(&audio_track);
return audio_track_result;
-
}
if(!gsr_audio_capture_add_track(&self->audio_capture, &audio_track)) {
+ gsr_audio_track_deinit(&audio_track);
return GSR_ERROR_GENERIC;
-
}
+
++audio_stream_index;
- if(audio_codec_context->frame_size > self->audio_max_frame_size)
- self->audio_max_frame_size = audio_codec_context->frame_size;
+ if(audio_track.codec_context->frame_size > self->audio_max_frame_size)
+ self->audio_max_frame_size = audio_track.codec_context->frame_size;
}
- //av_dump_format(self->av_format_context, 0, filename, 1);
-
return GSR_ERROR_OK;
}
@@ -768,8 +759,11 @@ int gsr_recorder_run(gsr_recorder *self) {
gsr_recording_clock_start(self->recording_clock);
self->record_start_time = gsr_recording_clock_get_start_time(self->recording_clock);
- if(gsr_audio_capture_start(&self->audio_capture, self->audio_max_frame_size, self->uses_amix) != GSR_ERROR_OK)
+ if(gsr_audio_capture_start(&self->audio_capture, self->audio_max_frame_size, self->uses_amix) != GSR_ERROR_OK) {
+ /* The audio threads that did start have to stop before they can be joined */
+ self->running = 0;
return GSR_ERROR_GENERIC;
+ }
// Set update_fps to 24 to test if duplicate/delayed frames cause video/audio desync or too fast/slow video.
//const double update_fps = fps + 190;
generated by cgit v1.2.3 (git 2.46.0) at 2026-08-22 21:04:28 +0000