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-rw-r--r--intern/ffmpeg/CMakeLists.txt1
-rw-r--r--intern/ffmpeg/ffmpeg_compat.h8
-rw-r--r--intern/ffmpeg/tests/ffmpeg_codecs.cc6
-rw-r--r--source/blender/blenkernel/intern/writeffmpeg.c48
4 files changed, 49 insertions, 14 deletions
diff --git a/intern/ffmpeg/CMakeLists.txt b/intern/ffmpeg/CMakeLists.txt
index 0de8496f3f3..4fb5df9d4cd 100644
--- a/intern/ffmpeg/CMakeLists.txt
+++ b/intern/ffmpeg/CMakeLists.txt
@@ -6,6 +6,7 @@ if(WITH_GTESTS)
tests/ffmpeg_codecs.cc
)
set(TEST_INC
+ .
)
set(TEST_INC_SYS
${FFMPEG_INCLUDE_DIRS}
diff --git a/intern/ffmpeg/ffmpeg_compat.h b/intern/ffmpeg/ffmpeg_compat.h
index f311e04d8e0..f7d87af8bca 100644
--- a/intern/ffmpeg/ffmpeg_compat.h
+++ b/intern/ffmpeg/ffmpeg_compat.h
@@ -36,6 +36,14 @@
# define FFMPEG_INLINE static inline
#endif
+#if (LIBAVFORMAT_VERSION_MAJOR < 59)
+/* For versions older than ffmpeg 5.0, use the old channel layout variables.
+ * We intend to only keep this workaround for around two releases (3.5, 3.6).
+ * If it sticks around any longer, then we should consider refactoring this.
+ */
+# define FFMPEG_USE_OLD_CHANNEL_VARS
+#endif
+
#if (LIBAVFORMAT_VERSION_MAJOR < 58) || \
((LIBAVFORMAT_VERSION_MAJOR == 58) && (LIBAVFORMAT_VERSION_MINOR < 76))
# define FFMPEG_USE_DURATION_WORKAROUND 1
diff --git a/intern/ffmpeg/tests/ffmpeg_codecs.cc b/intern/ffmpeg/tests/ffmpeg_codecs.cc
index 10cbe4b938b..cd06917f59b 100644
--- a/intern/ffmpeg/tests/ffmpeg_codecs.cc
+++ b/intern/ffmpeg/tests/ffmpeg_codecs.cc
@@ -3,6 +3,8 @@
#include "testing/testing.h"
extern "C" {
+#include "ffmpeg_compat.h"
+
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/log.h>
@@ -40,7 +42,11 @@ bool test_acodec(const AVCodec *codec, AVSampleFormat fmt)
if (ctx) {
ctx->sample_fmt = fmt;
ctx->sample_rate = 48000;
+#ifdef FFMPEG_USE_OLD_CHANNEL_VARS
+ ctx->channel_layout = AV_CH_LAYOUT_MONO;
+#else
av_channel_layout_from_mask(&ctx->ch_layout, AV_CH_LAYOUT_MONO);
+#endif
ctx->bit_rate = 128000;
int open = avcodec_open2(ctx, codec, NULL);
if (open >= 0) {
diff --git a/source/blender/blenkernel/intern/writeffmpeg.c b/source/blender/blenkernel/intern/writeffmpeg.c
index d71db8f71a5..4c11a2896a8 100644
--- a/source/blender/blenkernel/intern/writeffmpeg.c
+++ b/source/blender/blenkernel/intern/writeffmpeg.c
@@ -141,18 +141,25 @@ static int write_audio_frame(FFMpegContext *context)
frame->pts = context->audio_time / av_q2d(c->time_base);
frame->nb_samples = context->audio_input_samples;
frame->format = c->sample_fmt;
+# ifdef FFMPEG_USE_OLD_CHANNEL_VARS
+ frame->channels = c->channels;
+ frame->channel_layout = c->channel_layout;
+ const int num_channels = c->channels;
+# else
av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
+ const int num_channels = c->ch_layout.nb_channels;
+# endif
if (context->audio_deinterleave) {
int channel, i;
uint8_t *temp;
- for (channel = 0; channel < c->ch_layout.nb_channels; channel++) {
+ for (channel = 0; channel < num_channels; channel++) {
for (i = 0; i < frame->nb_samples; i++) {
memcpy(context->audio_deinterleave_buffer +
(i + channel * frame->nb_samples) * context->audio_sample_size,
context->audio_input_buffer +
- (c->ch_layout.nb_channels * i + channel) * context->audio_sample_size,
+ (num_channels * i + channel) * context->audio_sample_size,
context->audio_sample_size);
}
}
@@ -163,10 +170,10 @@ static int write_audio_frame(FFMpegContext *context)
}
avcodec_fill_audio_frame(frame,
- c->ch_layout.nb_channels,
+ num_channels,
c->sample_fmt,
context->audio_input_buffer,
- context->audio_input_samples * c->ch_layout.nb_channels *
+ context->audio_input_samples * num_channels *
context->audio_sample_size,
1);
@@ -944,25 +951,34 @@ static AVStream *alloc_audio_stream(FFMpegContext *context,
c->sample_rate = rd->ffcodecdata.audio_mixrate;
c->bit_rate = context->ffmpeg_audio_bitrate * 1000;
c->sample_fmt = AV_SAMPLE_FMT_S16;
- c->ch_layout.nb_channels = rd->ffcodecdata.audio_channels;
+ const int num_channels = rd->ffcodecdata.audio_channels;
+ int channel_layout_mask = 0;
switch (rd->ffcodecdata.audio_channels) {
case FFM_CHANNELS_MONO:
- av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_MONO);
+ channel_layout_mask = AV_CH_LAYOUT_MONO;
break;
case FFM_CHANNELS_STEREO:
- av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_STEREO);
+ channel_layout_mask = AV_CH_LAYOUT_STEREO;
break;
case FFM_CHANNELS_SURROUND4:
- av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_QUAD);
+ channel_layout_mask = AV_CH_LAYOUT_QUAD;
break;
case FFM_CHANNELS_SURROUND51:
- av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_5POINT1_BACK);
+ channel_layout_mask = AV_CH_LAYOUT_5POINT1_BACK;
break;
case FFM_CHANNELS_SURROUND71:
- av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_7POINT1);
+ channel_layout_mask = AV_CH_LAYOUT_7POINT1;
break;
}
+ BLI_assert(channel_layout_mask != 0);
+
+# ifdef FFMPEG_USE_OLD_CHANNEL_VARS
+ c->channels = num_channels;
+ c->channel_layout = channel_layout_mask;
+# else
+ av_channel_layout_from_mask(&c->ch_layout, channel_layout_mask);
+# endif
if (request_float_audio_buffer(codec_id)) {
/* mainly for AAC codec which is experimental */
@@ -1027,7 +1043,7 @@ static AVStream *alloc_audio_stream(FFMpegContext *context,
* not sure if that is needed anymore, so let's try out if there are any
* complaints regarding some FFmpeg versions users might have. */
context->audio_input_samples = AV_INPUT_BUFFER_MIN_SIZE * 8 / c->bits_per_coded_sample /
- c->ch_layout.nb_channels;
+ num_channels;
}
else {
context->audio_input_samples = c->frame_size;
@@ -1037,11 +1053,11 @@ static AVStream *alloc_audio_stream(FFMpegContext *context,
context->audio_sample_size = av_get_bytes_per_sample(c->sample_fmt);
- context->audio_input_buffer = (uint8_t *)av_malloc(
- context->audio_input_samples * c->ch_layout.nb_channels * context->audio_sample_size);
+ context->audio_input_buffer = (uint8_t *)av_malloc(context->audio_input_samples * num_channels *
+ context->audio_sample_size);
if (context->audio_deinterleave) {
context->audio_deinterleave_buffer = (uint8_t *)av_malloc(
- context->audio_input_samples * c->ch_layout.nb_channels * context->audio_sample_size);
+ context->audio_input_samples * num_channels * context->audio_sample_size);
}
context->audio_time = 0.0f;
@@ -1432,7 +1448,11 @@ int BKE_ffmpeg_start(void *context_v,
AVCodecContext *c = context->audio_codec;
AUD_DeviceSpecs specs;
+# ifdef FFMPEG_USE_OLD_CHANNEL_VARS
+ specs.channels = c->channels;
+# else
specs.channels = c->ch_layout.nb_channels;
+# endif
switch (av_get_packed_sample_fmt(c->sample_fmt)) {
case AV_SAMPLE_FMT_U8:
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-31 18:36:00 +0300