Bump ffmpeg to 5.1.2 "Riemann"

5 files changed, 14 insertions, 24 deletions

diff --git a/src/lib/audio_filter_graph.cc b/src/lib/audio_filter_graph.cc
index 4e3052d57..cb888c162 100644
--- a/src/lib/audio_filter_graph.cc
+++ b/src/lib/audio_filter_graph.cc
@@ -48,11 +48,7 @@ AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
 	/* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
 	   so we need to tell it we're using 16 channels if we are using more than 8.
 	*/
-	if (_channels > 8) {
-		_channel_layout = av_get_default_channel_layout (16);
-	} else {
-		_channel_layout = av_get_default_channel_layout (_channels);
-	}
+	av_channel_layout_default(&_channel_layout, _channels > 8 ? 16 : _channels);
 
 	_in_frame = av_frame_alloc ();
 	if (_in_frame == nullptr) {
@@ -69,7 +65,7 @@ string
 AudioFilterGraph::src_parameters () const
 {
 	char layout[64];
-	av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
+	av_channel_layout_describe(&_channel_layout, layout, sizeof(layout));
 
 	char buffer[256];
 	snprintf (
@@ -88,8 +84,9 @@ AudioFilterGraph::set_parameters (AVFilterContext* context) const
 	int r = av_opt_set_int_list (context, "sample_fmts", sample_fmts, AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
 	DCPOMATIC_ASSERT (r >= 0);
 
-	int64_t channel_layouts[] = { _channel_layout, -1 };
-	r = av_opt_set_int_list (context, "channel_layouts", channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
+	char ch_layout[64];
+	av_channel_layout_describe(&_channel_layout, ch_layout, sizeof(ch_layout));
+	r = av_opt_set(context, "ch_layouts", ch_layout, AV_OPT_SEARCH_CHILDREN);
 	DCPOMATIC_ASSERT (r >= 0);
 
 	int sample_rates[] = { _sample_rate, -1 };
@@ -114,7 +111,7 @@ void
 AudioFilterGraph::process (shared_ptr<AudioBuffers> buffers)
 {
 	DCPOMATIC_ASSERT (buffers->frames() > 0);
-	int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
+	int const process_channels = _channel_layout.nb_channels;
 	DCPOMATIC_ASSERT (process_channels >= buffers->channels());
 
 	if (buffers->channels() < process_channels) {
@@ -144,8 +141,10 @@ AudioFilterGraph::process (shared_ptr<AudioBuffers> buffers)
 	_in_frame->nb_samples = buffers->frames ();
 	_in_frame->format = AV_SAMPLE_FMT_FLTP;
 	_in_frame->sample_rate = _sample_rate;
-	_in_frame->channel_layout = _channel_layout;
+	_in_frame->ch_layout = _channel_layout;
+LIBDCP_DISABLE_WARNINGS
 	_in_frame->channels = process_channels;
+LIBDCP_ENABLE_WARNINGS
 
 	int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
 
diff --git a/src/lib/audio_filter_graph.h b/src/lib/audio_filter_graph.h
index e5c55fa27..1ac2b071b 100644
--- a/src/lib/audio_filter_graph.h
+++ b/src/lib/audio_filter_graph.h
@@ -47,7 +47,7 @@ protected:
 private:
 	int _sample_rate;
 	int _channels;
-	int64_t _channel_layout;
+	AVChannelLayout _channel_layout;
 	AVFrame* _in_frame;
 };
 
diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc
index 0d43156c0..3925853c9 100644
--- a/src/lib/ffmpeg_decoder.cc
+++ b/src/lib/ffmpeg_decoder.cc
@@ -260,7 +260,7 @@ deinterleave_audio(AVFrame* frame)
 
 	/* XXX: can't we use swr_convert() to do the format conversion? */
 
-	int const channels = frame->channels;
+	int const channels = frame->ch_layout.nb_channels;
 	int const frames = frame->nb_samples;
 	int const total_samples = frames * channels;
 	auto audio = make_shared<AudioBuffers>(channels, frames);
diff --git a/src/lib/ffmpeg_examiner.cc b/src/lib/ffmpeg_examiner.cc
index ed6f21094..d2d1093a9 100644
--- a/src/lib/ffmpeg_examiner.cc
+++ b/src/lib/ffmpeg_examiner.cc
@@ -73,14 +73,6 @@ FFmpegExaminer::FFmpegExaminer (shared_ptr<const FFmpegContent> c, shared_ptr<Jo
 		auto codec = _codec_context[i] ? _codec_context[i]->codec : nullptr;
 		if (s->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && codec) {
 
-			/* This is a hack; sometimes it seems that _audio_codec_context->channel_layout isn't set up,
-			   so bodge it here.  No idea why we should have to do this.
-			*/
-
-			if (s->codecpar->channel_layout == 0) {
-				s->codecpar->channel_layout = av_get_default_channel_layout (s->codecpar->channels);
-			}
-
 			DCPOMATIC_ASSERT (_format_context->duration != AV_NOPTS_VALUE);
 			DCPOMATIC_ASSERT (codec->name);
 
@@ -91,7 +83,7 @@ FFmpegExaminer::FFmpegExaminer (shared_ptr<const FFmpegContent> c, shared_ptr<Jo
 					s->id,
 					s->codecpar->sample_rate,
 					llrint ((double(_format_context->duration) / AV_TIME_BASE) * s->codecpar->sample_rate),
-					s->codecpar->channels,
+					s->codecpar->ch_layout.nb_channels,
 					s->codecpar->bits_per_raw_sample ? s->codecpar->bits_per_raw_sample : s->codecpar->bits_per_coded_sample
 					)
 				);
diff --git a/src/lib/ffmpeg_file_encoder.cc b/src/lib/ffmpeg_file_encoder.cc
index 6d1ad68f7..d7833265d 100644
--- a/src/lib/ffmpeg_file_encoder.cc
+++ b/src/lib/ffmpeg_file_encoder.cc
@@ -73,8 +73,7 @@ public:
 		_codec_context->bit_rate = channels * 128 * 1024;
 		_codec_context->sample_fmt = sample_format;
 		_codec_context->sample_rate = frame_rate;
-		_codec_context->channel_layout = av_get_default_channel_layout (channels);
-		_codec_context->channels = channels;
+		av_channel_layout_default(&_codec_context->ch_layout, channels);
 
 		int r = avcodec_open2 (_codec_context, _codec, 0);
 		if (r < 0) {
@@ -143,7 +142,7 @@ public:
 
 		frame->nb_samples = size;
 		frame->format = _codec_context->sample_fmt;
-		frame->channels = channels;
+		frame->ch_layout.nb_channels = channels;
 		int r = avcodec_fill_audio_frame (frame, channels, _codec_context->sample_fmt, (const uint8_t *) samples, buffer_size, 0);
 		DCPOMATIC_ASSERT (r >= 0);