diff options
Diffstat (limited to 'source/blender/blenkernel/intern/writeffmpeg.c')
| -rw-r--r-- | source/blender/blenkernel/intern/writeffmpeg.c | 42 |
1 files changed, 21 insertions, 21 deletions
diff --git a/source/blender/blenkernel/intern/writeffmpeg.c b/source/blender/blenkernel/intern/writeffmpeg.c index 0d3a790ba00..d71db8f71a5 100644 --- a/source/blender/blenkernel/intern/writeffmpeg.c +++ b/source/blender/blenkernel/intern/writeffmpeg.c @@ -141,19 +141,18 @@ 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; - frame->channels = c->channels; - frame->channel_layout = c->channel_layout; + av_channel_layout_copy(&frame->ch_layout, &c->ch_layout); if (context->audio_deinterleave) { int channel, i; uint8_t *temp; - for (channel = 0; channel < c->channels; channel++) { + for (channel = 0; channel < c->ch_layout.nb_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->channels * i + channel) * context->audio_sample_size, + (c->ch_layout.nb_channels * i + channel) * context->audio_sample_size, context->audio_sample_size); } } @@ -164,10 +163,11 @@ static int write_audio_frame(FFMpegContext *context) } avcodec_fill_audio_frame(frame, - c->channels, + c->ch_layout.nb_channels, c->sample_fmt, context->audio_input_buffer, - context->audio_input_samples * c->channels * context->audio_sample_size, + context->audio_input_samples * c->ch_layout.nb_channels * + context->audio_sample_size, 1); int success = 1; @@ -492,7 +492,7 @@ static const AVCodec *get_av1_encoder( /* Apply AV1 encoder specific settings. */ if (codec) { - if (strcmp(codec->name, "librav1e") == 0) { + if (STREQ(codec->name, "librav1e")) { /* Set "tiles" to 8 to enable multi-threaded encoding. */ if (rd->threads > 8) { ffmpeg_dict_set_int(opts, "tiles", rd->threads); @@ -530,7 +530,7 @@ static const AVCodec *get_av1_encoder( BLI_snprintf(buffer, sizeof(buffer), "keyint=%d", context->ffmpeg_gop_size); av_dict_set(opts, "rav1e-params", buffer, 0); } - else if (strcmp(codec->name, "libsvtav1") == 0) { + else if (STREQ(codec->name, "libsvtav1")) { /* Set preset value based on ffmpeg_preset. * Must check context->ffmpeg_preset again in case this encoder was selected due to the * absence of another. */ @@ -552,7 +552,7 @@ static const AVCodec *get_av1_encoder( ffmpeg_dict_set_int(opts, "qp", context->ffmpeg_crf); } } - else if (strcmp(codec->name, "libaom-av1") == 0) { + else if (STREQ(codec->name, "libaom-av1")) { /* Speed up libaom-av1 encoding by enabling multithreading and setting tiles. */ ffmpeg_dict_set_int(opts, "row-mt", 1); const char *tiles_string = NULL; @@ -944,23 +944,23 @@ 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->channels = rd->ffcodecdata.audio_channels; + c->ch_layout.nb_channels = rd->ffcodecdata.audio_channels; switch (rd->ffcodecdata.audio_channels) { case FFM_CHANNELS_MONO: - c->channel_layout = AV_CH_LAYOUT_MONO; + av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_MONO); break; case FFM_CHANNELS_STEREO: - c->channel_layout = AV_CH_LAYOUT_STEREO; + av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_STEREO); break; case FFM_CHANNELS_SURROUND4: - c->channel_layout = AV_CH_LAYOUT_QUAD; + av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_QUAD); break; case FFM_CHANNELS_SURROUND51: - c->channel_layout = AV_CH_LAYOUT_5POINT1_BACK; + av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_5POINT1_BACK); break; case FFM_CHANNELS_SURROUND71: - c->channel_layout = AV_CH_LAYOUT_7POINT1; + av_channel_layout_from_mask(&c->ch_layout, AV_CH_LAYOUT_7POINT1); break; } @@ -1017,7 +1017,7 @@ static AVStream *alloc_audio_stream(FFMpegContext *context, return NULL; } - /* need to prevent floating point exception when using vorbis audio codec, + /* Need to prevent floating point exception when using VORBIS audio codec, * initialize this value in the same way as it's done in FFmpeg itself (sergey) */ c->time_base.num = 1; c->time_base.den = c->sample_rate; @@ -1027,7 +1027,7 @@ 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->channels; + c->ch_layout.nb_channels; } else { context->audio_input_samples = c->frame_size; @@ -1037,11 +1037,11 @@ 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->channels * - context->audio_sample_size); + context->audio_input_buffer = (uint8_t *)av_malloc( + context->audio_input_samples * c->ch_layout.nb_channels * context->audio_sample_size); if (context->audio_deinterleave) { context->audio_deinterleave_buffer = (uint8_t *)av_malloc( - context->audio_input_samples * c->channels * context->audio_sample_size); + context->audio_input_samples * c->ch_layout.nb_channels * context->audio_sample_size); } context->audio_time = 0.0f; @@ -1432,7 +1432,7 @@ int BKE_ffmpeg_start(void *context_v, AVCodecContext *c = context->audio_codec; AUD_DeviceSpecs specs; - specs.channels = c->channels; + specs.channels = c->ch_layout.nb_channels; switch (av_get_packed_sample_fmt(c->sample_fmt)) { case AV_SAMPLE_FMT_U8: |