num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
- decklink_output_surface(create_surface(format)),
- audio_mixer(num_cards)
+ decklink_output_surface(create_surface(format))
{
memcpy(ycbcr_interpretation, global_flags.ycbcr_interpretation, sizeof(ycbcr_interpretation));
CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
// Must be instantiated after VideoEncoder has initialized global_flags.use_zerocopy.
theme.reset(new Theme(global_flags.theme_filename, global_flags.theme_dirs, resource_pool.get(), num_cards));
+ // Must be instantiated after the theme, as the theme decides the number of FFmpeg inputs.
+ std::vector<FFmpegCapture *> video_inputs = theme->get_video_inputs();
+ audio_mixer.reset(new AudioMixer(num_cards, video_inputs.size()));
+
httpd.add_endpoint("/channels", bind(&Mixer::get_channels_json, this), HTTPD::ALLOW_ALL_ORIGINS);
for (int channel_idx = 2; channel_idx < theme->get_num_channels(); ++channel_idx) {
char url[256];
// Initialize all video inputs the theme asked for. Note that these are
// all put _after_ the regular cards, which stop at <num_cards> - 1.
- std::vector<FFmpegCapture *> video_inputs = theme->get_video_inputs();
for (unsigned video_card_index = 0; video_card_index < video_inputs.size(); ++card_index, ++video_card_index) {
if (card_index >= MAX_VIDEO_CARDS) {
fprintf(stderr, "ERROR: Not enough card slots available for the videos the theme requested.\n");
// NOTE: start_bm_capture() happens in thread_func().
- DeviceSpec device{InputSourceType::CAPTURE_CARD, card_index};
- audio_mixer.reset_resampler(device);
- audio_mixer.set_display_name(device, card->capture->get_description());
- audio_mixer.trigger_state_changed_callback();
+ DeviceSpec device;
+ if (card_type == CardType::FFMPEG_INPUT) {
+ device = DeviceSpec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index - num_cards};
+ } else {
+ device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
+ }
+ audio_mixer->reset_resampler(device);
+ audio_mixer->set_display_name(device, card->capture->get_description());
+ audio_mixer->trigger_state_changed_callback();
// Unregister old metrics, if any.
if (!card->labels.empty()) {
FrameAllocator::Frame video_frame, size_t video_offset, VideoFormat video_format,
FrameAllocator::Frame audio_frame, size_t audio_offset, AudioFormat audio_format)
{
- DeviceSpec device{InputSourceType::CAPTURE_CARD, card_index};
+ DeviceSpec device;
+ if (card_index >= num_cards) {
+ device = DeviceSpec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index - num_cards};
+ } else {
+ device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
+ }
CaptureCard *card = &cards[card_index];
++card->metric_input_received_frames;
assert(frame_length > 0);
size_t num_samples = (audio_frame.len > audio_offset) ? (audio_frame.len - audio_offset) / audio_format.num_channels / (audio_format.bits_per_sample / 8) : 0;
- if (num_samples > OUTPUT_FREQUENCY / 10) {
+ if (num_samples > OUTPUT_FREQUENCY / 10 && card->type != CardType::FFMPEG_INPUT) {
printf("%s: Dropping frame with implausible audio length (len=%d, offset=%d) [timecode=0x%04x video_len=%d video_offset=%d video_format=%x)\n",
spec_to_string(device).c_str(), int(audio_frame.len), int(audio_offset),
timecode, int(video_frame.len), int(video_offset), video_format.id);
if (dropped_frames > MAX_FPS * 2) {
fprintf(stderr, "%s lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
spec_to_string(device).c_str(), card->last_timecode, timecode);
- audio_mixer.reset_resampler(device);
+ audio_mixer->reset_resampler(device);
dropped_frames = 0;
++card->metric_input_resets;
} else if (dropped_frames > 0) {
bool success;
do {
- success = audio_mixer.add_silence(device, silence_samples, dropped_frames, frame_length);
+ success = audio_mixer->add_silence(device, silence_samples, dropped_frames, frame_length);
} while (!success);
}
if (num_samples > 0) {
- audio_mixer.add_audio(device, audio_frame.data + audio_offset, num_samples, audio_format, frame_length, audio_frame.received_timestamp);
+ audio_mixer->add_audio(device, audio_frame.data + audio_offset, num_samples, audio_format, frame_length, audio_frame.received_timestamp);
}
// Done with the audio, so release it.
ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
task.adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
- vector<float> samples_out = audio_mixer.get_output(
+ vector<float> samples_out = audio_mixer->get_output(
task.frame_timestamp,
task.num_samples,
rate_adjustment_policy);