libobs: Add surround sound audio support
(This commit also modifies the following modules: UI, deps/media-playback, coreaudio-encoder, decklink, linux-alsa, linux-pulseaudio, mac-capture, obs-ffmpeg, obs-filters, obs-libfdk, obs-outputs, win-dshow, and win-wasapi) Adds surround sound audio support to the core, core plugins, and user interface. Compatible streaming services: Twitch, FB 360 live Compatible protocols: rtmp / mpeg-ts tcp udp Compatible file formats: mkv mp4 ts (others untested) Compatible codecs: ffmpeg aac, fdk_aac, CoreAudio aac, opus, vorbis, pcm (others untested). Tested streaming servers: wowza, nginx HLS, mpeg-dash : surround passthrough Html5 players tested with live surround: videojs, mediaelement, viblast (hls+dash), hls.js Decklink: on win32, swap channels order for 5.1 7.1 (due to different channel mapping on wav, mpeg, ffmpeg) Audio filters: surround working. Monitoring: surround working (win macOs linux (pulse-audio)). VST: stereo plugins keep in general only the first two channels. surround plugins should work (e.g. mcfx does). OS: win, macOs, linux (alsa, pulse-audio). Misc: larger audio bitrates unlocked to accommodate more channels NB: mf-aac only supports mono and stereo + 5.1 on win 10 (not implemented due to lack of usefulness) Closes jp9000/obs-studio#968
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@@ -56,6 +56,45 @@ struct enc_encoder {
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int frame_size_bytes;
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};
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static inline uint64_t convert_speaker_layout(enum speaker_layout layout)
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{
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switch (layout) {
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case SPEAKERS_UNKNOWN: return 0;
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case SPEAKERS_MONO: return AV_CH_LAYOUT_MONO;
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case SPEAKERS_STEREO: return AV_CH_LAYOUT_STEREO;
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case SPEAKERS_2POINT1: return AV_CH_LAYOUT_2_1;
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case SPEAKERS_QUAD: return AV_CH_LAYOUT_QUAD;
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case SPEAKERS_4POINT1: return AV_CH_LAYOUT_4POINT1;
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case SPEAKERS_5POINT1: return AV_CH_LAYOUT_5POINT1;
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case SPEAKERS_5POINT1_SURROUND: return AV_CH_LAYOUT_5POINT1_BACK;
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case SPEAKERS_7POINT1: return AV_CH_LAYOUT_7POINT1;
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case SPEAKERS_7POINT1_SURROUND: return AV_CH_LAYOUT_7POINT1_WIDE_BACK;
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case SPEAKERS_SURROUND: return AV_CH_LAYOUT_SURROUND;
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}
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/* shouldn't get here */
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return 0;
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}
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static inline enum speaker_layout convert_ff_channel_layout(uint64_t channel_layout)
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{
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switch (channel_layout) {
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case AV_CH_LAYOUT_MONO: return SPEAKERS_MONO;
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case AV_CH_LAYOUT_STEREO: return SPEAKERS_STEREO;
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case AV_CH_LAYOUT_2_1: return SPEAKERS_2POINT1;
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case AV_CH_LAYOUT_QUAD: return SPEAKERS_QUAD;
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case AV_CH_LAYOUT_4POINT1: return SPEAKERS_4POINT1;
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case AV_CH_LAYOUT_5POINT1: return SPEAKERS_5POINT1;
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case AV_CH_LAYOUT_5POINT1_BACK: return SPEAKERS_5POINT1_SURROUND;
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case AV_CH_LAYOUT_7POINT1: return SPEAKERS_7POINT1;
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case AV_CH_LAYOUT_7POINT1_WIDE_BACK: return SPEAKERS_7POINT1_SURROUND;
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case AV_CH_LAYOUT_SURROUND: return SPEAKERS_SURROUND;
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}
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/* shouldn't get here */
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return SPEAKERS_UNKNOWN;
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}
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static const char *aac_getname(void *unused)
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{
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UNUSED_PARAMETER(unused);
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@@ -169,7 +208,10 @@ static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
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}
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enc->context->bit_rate = bitrate * 1000;
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const struct audio_output_info *aoi;
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aoi = audio_output_get_info(audio);
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enc->context->channels = (int)audio_output_get_channels(audio);
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enc->context->channel_layout = convert_speaker_layout(aoi->speakers);
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enc->context->sample_rate = audio_output_get_sample_rate(audio);
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enc->context->sample_fmt = enc->codec->sample_fmts ?
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enc->codec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
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@@ -206,8 +248,9 @@ static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
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enc->context->cutoff = cutoff;
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}
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info("bitrate: %" PRId64 ", channels: %d",
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enc->context->bit_rate / 1000, enc->context->channels);
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info("bitrate: %" PRId64 ", channels: %d, channel_layout: %x\n",
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enc->context->bit_rate / 1000, enc->context->channels,
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enc->context->channel_layout);
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init_sizes(enc, audio);
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@@ -303,9 +346,8 @@ static obs_properties_t *enc_properties(void *unused)
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UNUSED_PARAMETER(unused);
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obs_properties_t *props = obs_properties_create();
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obs_properties_add_int(props, "bitrate",
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obs_module_text("Bitrate"), 64, 320, 32);
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obs_module_text("Bitrate"), 64, 1024, 32);
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return props;
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}
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@@ -323,6 +365,7 @@ static void enc_audio_info(void *data, struct audio_convert_info *info)
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struct enc_encoder *enc = data;
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info->format = convert_ffmpeg_sample_format(enc->context->sample_fmt);
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info->samples_per_sec = (uint32_t)enc->context->sample_rate;
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info->speakers = convert_ff_channel_layout(enc->context->channel_layout);
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}
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static size_t enc_frame_size(void *data)
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