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https://github.com/obsproject/obs-studio.git
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obs-ffmpeg: Fix for channel layout API change
The channel_layout API was overhauled by FFmpeg [1-4]. The previous bitmask channel_layout is replaced by a struct ch_layout which combines the number of channels, a bitmask and other infos. This struct must now be supplied to AVframes since avutil >= 57.24.100 and to AVCodecContext since avcodec 59.24.100 per (1]. This commit provides the required info to ffmpeg-mux, obs-ffmpeg-output & to obs-ffmpeg-audio-encoders. [1] Bump minor versions after the channel layout changes https://github.com/FFmpeg/FFmpeg/commit/cdba98bb80e2ab73d34659c610771b020afc6a77 [2] lavc: switch to the new channel layout API https://github.com/FFmpeg/FFmpeg/commit/548aeb93834b8425c86d1ce60fddc1d41805724d [3] avutil/channel_layout: Add a new channel layout API https://github.com/FFmpeg/FFmpeg/commit/086a8048061bf9fb4c63943f6962db48175f655c [4] avframe: switch to the new channel layout API db6efa18 https://github.com/FFmpeg/FFmpeg/commit/db6efa1815e217ed76f39aee8b15ee5c64698537 Signed-off-by: pkv <pkv@obsproject.com>
This commit is contained in:
@@ -495,6 +495,7 @@ static void create_audio_stream(struct ffmpeg_mux *ffm, int idx)
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context->time_base = stream->time_base;
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context->extradata = extradata;
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context->extradata_size = ffm->audio_header[idx].size;
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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context->channel_layout =
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av_get_default_channel_layout(context->channels);
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//avutil default channel layout for 4 channels is 4.0 ; fix for quad
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@@ -503,6 +504,12 @@ static void create_audio_stream(struct ffmpeg_mux *ffm, int idx)
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//avutil default channel layout for 5 channels is 5.0 ; fix for 4.1
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if (context->channels == 5)
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context->channel_layout = av_get_channel_layout("4.1");
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#else
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av_channel_layout_default(&context->ch_layout, context->channels);
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//avutil default channel layout for 5 channels is 5.0 ; fix for 4.1
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if (context->channels == 5)
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context->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_4POINT1;
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#endif
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if (ffm->output->oformat->flags & AVFMT_GLOBALHEADER)
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context->flags |= CODEC_FLAG_GLOBAL_H;
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@@ -57,6 +57,7 @@ struct enc_encoder {
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int frame_size_bytes;
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};
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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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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@@ -81,30 +82,7 @@ static inline uint64_t convert_speaker_layout(enum speaker_layout layout)
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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
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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:
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return SPEAKERS_MONO;
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case AV_CH_LAYOUT_STEREO:
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return SPEAKERS_STEREO;
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case AV_CH_LAYOUT_SURROUND:
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return SPEAKERS_2POINT1;
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case AV_CH_LAYOUT_4POINT0:
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return SPEAKERS_4POINT0;
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case AV_CH_LAYOUT_4POINT1:
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return SPEAKERS_4POINT1;
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case AV_CH_LAYOUT_5POINT1_BACK:
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return SPEAKERS_5POINT1;
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case AV_CH_LAYOUT_7POINT1:
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return SPEAKERS_7POINT1;
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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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#endif
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static const char *aac_getname(void *unused)
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{
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@@ -157,7 +135,11 @@ static bool initialize_codec(struct enc_encoder *enc)
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}
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enc->aframe->format = enc->context->sample_fmt;
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enc->aframe->channels = enc->context->channels;
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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enc->aframe->channel_layout = enc->context->channel_layout;
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#else
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enc->aframe->ch_layout = enc->context->ch_layout;
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#endif
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enc->aframe->sample_rate = enc->context->sample_rate;
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enc->frame_size = enc->context->frame_size;
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@@ -235,7 +217,15 @@ static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
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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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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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enc->context->channel_layout = convert_speaker_layout(aoi->speakers);
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#else
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av_channel_layout_default(&enc->context->ch_layout,
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enc->context->channels);
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if (aoi->speakers == SPEAKERS_4POINT1)
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enc->context->ch_layout =
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(AVChannelLayout)AV_CHANNEL_LAYOUT_4POINT1;
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#endif
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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]
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@@ -264,11 +254,18 @@ static void *enc_create(obs_data_t *settings, obs_encoder_t *encoder,
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av_opt_set(enc->context->priv_data, "aac_coder", "fast", 0);
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}
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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info("bitrate: %" PRId64 ", channels: %d, channel_layout: %x\n",
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(int64_t)enc->context->bit_rate / 1000,
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(int)enc->context->channels,
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(unsigned int)enc->context->channel_layout);
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#else
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char buf[256];
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av_channel_layout_describe(&enc->context->ch_layout, buf, 256);
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info("bitrate: %" PRId64 ", channels: %d, channel_layout: %s\n",
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(int64_t)enc->context->bit_rate / 1000,
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(int)enc->context->channels, buf);
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#endif
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init_sizes(enc, audio);
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/* enable experimental FFmpeg encoder if the only one available */
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@@ -306,7 +303,9 @@ static bool do_encode(struct enc_encoder *enc, struct encoder_packet *packet,
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enc->aframe->pts = av_rescale_q(
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enc->total_samples, (AVRational){1, enc->context->sample_rate},
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enc->context->time_base);
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#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 24, 100)
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enc->aframe->ch_layout = enc->context->ch_layout;
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#endif
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ret = avcodec_fill_audio_frame(
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enc->aframe, enc->context->channels, enc->context->sample_fmt,
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enc->samples[0], enc->frame_size_bytes * enc->context->channels,
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@@ -394,8 +393,10 @@ 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 =
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convert_ff_channel_layout(enc->context->channel_layout);
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if (enc->context->channels != 7 && enc->context->channels <= 8)
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info->speakers = (enum speaker_layout)(enc->context->channels);
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else
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info->speakers = SPEAKERS_UNKNOWN;
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}
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static size_t enc_frame_size(void *data)
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@@ -321,9 +321,12 @@ static bool open_audio_codec(struct ffmpeg_data *data, int idx)
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data->aframe[idx]->format = context->sample_fmt;
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data->aframe[idx]->channels = context->channels;
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#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 24, 100)
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data->aframe[idx]->channel_layout = context->channel_layout;
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#else
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data->aframe[idx]->ch_layout = context->ch_layout;
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#endif
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data->aframe[idx]->sample_rate = context->sample_rate;
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context->strict_std_compliance = -2;
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ret = avcodec_open2(context, data->acodec, NULL);
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@@ -378,13 +381,18 @@ static bool create_audio_stream(struct ffmpeg_data *data, int idx)
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context->time_base = (AVRational){1, aoi.samples_per_sec};
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context->channels = get_audio_channels(aoi.speakers);
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context->sample_rate = aoi.samples_per_sec;
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(59, 24, 100)
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context->channel_layout =
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av_get_default_channel_layout(context->channels);
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//avutil default channel layout for 5 channels is 5.0 ; fix for 4.1
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if (aoi.speakers == SPEAKERS_4POINT1)
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context->channel_layout = av_get_channel_layout("4.1");
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#else
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av_channel_layout_default(&context->ch_layout, context->channels);
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if (aoi.speakers == SPEAKERS_4POINT1)
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context->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_4POINT1;
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#endif
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context->sample_fmt = data->acodec->sample_fmts
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? data->acodec->sample_fmts[0]
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: AV_SAMPLE_FMT_FLTP;
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