X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=133fc28108a9b50e68e1e7ffa46848313a5e3d65;hb=refs%2Ftags%2F1.0.0;hp=cfdb5b190b1b25f828227d6b59fdc8fe9eeda455;hpb=593173ffcdbccff777720b3c41c3e771fa93a10a;p=nageru diff --git a/mixer.cpp b/mixer.cpp index cfdb5b1..133fc28 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -256,7 +256,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, decode_video_format(video_format, &width, &height, &second_field_start, &extra_lines_top, &extra_lines_bottom, &frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now. - int64_t frame_length = TIMEBASE * frame_rate_den / frame_rate_nom; + int64_t frame_length = int64_t(TIMEBASE * frame_rate_den) / frame_rate_nom; size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0; if (num_samples > OUTPUT_FREQUENCY / 10) { @@ -510,6 +510,7 @@ void Mixer::thread_func() } // Resample the audio as needed, including from previously dropped frames. + assert(num_cards > 0); for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) { { // Signal to the audio thread to process this frame. @@ -691,6 +692,11 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) { vector samples_card; vector samples_out; + + // TODO: Allow mixing audio from several sources. + unsigned selected_audio_card = theme->map_signal(audio_source_channel); + assert(selected_audio_card < num_cards); + for (unsigned card_index = 0; card_index < num_cards; ++card_index) { samples_card.resize(num_samples * 2); { @@ -699,8 +705,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) printf("Card %d reported previous underrun.\n", card_index); } } - // TODO: Allow using audio from the other card(s) as well. - if (card_index == 0) { + if (card_index == selected_audio_card) { samples_out = move(samples_card); } } @@ -784,6 +789,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) peak_resampler.process(); size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count; peak = max(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2)); + peak_resampler.out_data = nullptr; } // At this point, we are most likely close to +0 LU, but all of our