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commit c31a781412a3aa801cc0997ba001177f258f4d80 Author: James Almer <[email protected]> AuthorDate: Sun Jul 5 11:12:01 2026 -0300 Commit: James Almer <[email protected]> CommitDate: Sat Jul 11 11:10:12 2026 -0300 avcodec/audio_frame_queue: add skip samples side data to output packets Instead of reimplementing the same code on every afq user, just do it here. Signed-off-by: James Almer <[email protected]> --- libavcodec/aacenc.c | 14 +++----------- libavcodec/ac3enc.c | 15 +++------------ libavcodec/aptxenc.c | 5 ++++- libavcodec/audio_frame_queue.c | 29 +++++++++++++++++++++++------ libavcodec/audio_frame_queue.h | 8 ++++---- libavcodec/audiotoolboxenc.c | 7 ++++--- libavcodec/libfdk-aacenc.c | 30 +++++------------------------- libavcodec/libmp3lame.c | 26 ++++---------------------- libavcodec/libopencore-amr.c | 5 +++-- libavcodec/libopusenc.c | 19 +++---------------- libavcodec/libshine.c | 5 +++-- libavcodec/libspeexenc.c | 6 ++++-- libavcodec/libvorbisenc.c | 16 +++------------- libavcodec/mlpenc.c | 7 +++---- libavcodec/nellymoserenc.c | 5 +++-- libavcodec/opus/enc.c | 4 +++- libavcodec/ra144enc.c | 5 +++-- libavcodec/vorbisenc.c | 4 +++- tests/ref/fate/ac3-fixed-encode-2 | 2 +- tests/ref/fate/ac3-fixed-encode-3 | 2 +- tests/ref/fate/copy-shortest1 | 2 +- tests/ref/fate/copy-shortest2 | 2 +- tests/ref/fate/ffmpeg-filter_complex_audio | 2 +- tests/ref/fate/opus-enc-silence | 2 +- tests/ref/fate/shortest | 2 +- 25 files changed, 88 insertions(+), 136 deletions(-) diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 88d5ff55f7..6fc9254ae8 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -1412,17 +1412,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, s->lambda_sum += (s->nmr && s->nmr->lam_rc > 0.0f) ? s->nmr->lam_rc : s->lambda; s->lambda_count++; - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); - - int discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration); - if (discard_padding > 0) { - uint8_t *side_data = - av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10); - if (!side_data) - return AVERROR(ENOMEM); - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; avpkt->flags |= AV_PKT_FLAG_KEY; diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c index 69744989ba..0482d9ee58 100644 --- a/libavcodec/ac3enc.c +++ b/libavcodec/ac3enc.c @@ -1982,7 +1982,6 @@ int ff_ac3_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { AC3EncodeContext *const s = avctx->priv_data; - int discard_padding; int ret; /* add current frame to queue */ @@ -2025,17 +2024,9 @@ int ff_ac3_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, return ret; ac3_output_frame(s, avpkt->data); - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); - - discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration); - if (discard_padding > 0) { - uint8_t *side_data = - av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10); - if (!side_data) - return AVERROR(ENOMEM); - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; return 0; diff --git a/libavcodec/aptxenc.c b/libavcodec/aptxenc.c index ab02459733..1799befee8 100644 --- a/libavcodec/aptxenc.c +++ b/libavcodec/aptxenc.c @@ -240,7 +240,10 @@ static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, aptx_encode_samples(s, samples, avpkt->data + pos); } - ff_af_queue_remove(&s0->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration); + ret = ff_af_queue_remove(&s0->afq, frame->nb_samples, avpkt); + if (ret < 0) + return ret; + *got_packet_ptr = 1; return 0; } diff --git a/libavcodec/audio_frame_queue.c b/libavcodec/audio_frame_queue.c index 1db4a35673..2023c92095 100644 --- a/libavcodec/audio_frame_queue.c +++ b/libavcodec/audio_frame_queue.c @@ -31,6 +31,7 @@ av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq) afq->avctx = avctx; afq->remaining_delay = avctx->initial_padding; afq->remaining_samples = avctx->initial_padding; + afq->output_delay = avctx->initial_padding; afq->frame_count = 0; } @@ -83,10 +84,10 @@ int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f) return 0; } -void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, - int64_t *duration) +int ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, AVPacket *pkt) { int64_t out_pts = AV_NOPTS_VALUE; + int frame_size = nb_samples; int removed_samples = 0; int i; @@ -96,8 +97,8 @@ void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, } if(!afq->frame_count) av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but the queue is empty\n", nb_samples); - if (pts) - *pts = ff_samples_to_time_base(afq->avctx, out_pts); + if (pkt) + pkt->pts = ff_samples_to_time_base(afq->avctx, out_pts); for(i=0; nb_samples && i<afq->frame_count; i++){ int n= FFMIN(afq->frames[i].duration, nb_samples); @@ -119,6 +120,22 @@ void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, afq->frames[0].pts += nb_samples; av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than there are in the queue\n", nb_samples); } - if (duration) - *duration = ff_samples_to_time_base(afq->avctx, removed_samples); + if (pkt) { + int discard_padding = frame_size - removed_samples; + pkt->duration = ff_samples_to_time_base(afq->avctx, removed_samples); + if (afq->output_delay > 0 || discard_padding > 0) { + uint8_t *side_data = + av_packet_new_side_data(pkt, AV_PKT_DATA_SKIP_SAMPLES, 10); + if (!side_data) + return AVERROR(ENOMEM); + if (afq->output_delay) { + AV_WL32(side_data, FFMIN(afq->output_delay, removed_samples)); + afq->output_delay -= removed_samples; + afq->output_delay = FFMAX(afq->output_delay, 0); + } + AV_WL32(side_data + 4, discard_padding); + } + } + + return 0; } diff --git a/libavcodec/audio_frame_queue.h b/libavcodec/audio_frame_queue.h index d8076eae54..62f80fb80f 100644 --- a/libavcodec/audio_frame_queue.h +++ b/libavcodec/audio_frame_queue.h @@ -31,6 +31,7 @@ typedef struct AudioFrame { typedef struct AudioFrameQueue { AVCodecContext *avctx; + int output_delay; int remaining_delay; int remaining_samples; AudioFrame *frames; @@ -74,10 +75,9 @@ int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f); * * @param afq queue context * @param nb_samples number of samples to remove from the queue - * @param[out] pts output packet pts - * @param[out] duration output packet duration + * @param[out] pkt output packet + * @return 0 on success, negative AVERROR code on failure */ -void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, - int64_t *duration); +int ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, AVPacket *pkt); #endif /* AVCODEC_AUDIO_FRAME_QUEUE_H */ diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c index d999e5e8a2..11af7c0154 100644 --- a/libavcodec/audiotoolboxenc.c +++ b/libavcodec/audiotoolboxenc.c @@ -571,11 +571,12 @@ static int ffat_encode(AVCodecContext *avctx, AVPacket *avpkt, if ((!ret || ret == 1) && *got_packet_ptr) { avpkt->size = out_buffers.mBuffers[0].mDataByteSize; - ff_af_queue_remove(&at->afq, out_pkt_desc.mVariableFramesInPacket ? + int ret = ff_af_queue_remove(&at->afq, out_pkt_desc.mVariableFramesInPacket ? out_pkt_desc.mVariableFramesInPacket : avctx->frame_size, - &avpkt->pts, - &avpkt->duration); + avpkt); + if (ret < 0) + return ret; avpkt->flags |= AV_PKT_FLAG_KEY; } else if (ret && ret != 1) { av_log(avctx, AV_LOG_ERROR, "Encode error: %i\n", ret); diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c index 168b9d775d..033cf1f33f 100644 --- a/libavcodec/libfdk-aacenc.c +++ b/libavcodec/libfdk-aacenc.c @@ -141,7 +141,6 @@ static void aac_encode_flush(AVCodecContext *avctx) AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; AACENC_InArgs in_args = { 0 }; AACENC_OutArgs out_args; - int64_t pts, duration; uint8_t dummy_in[1], dummy_out[1]; int in_buffer_identifiers[] = { IN_AUDIO_DATA, IN_METADATA_SETUP }; int in_buffer_element_sizes[] = { 2, sizeof(AACENC_MetaData) }; @@ -152,7 +151,7 @@ static void aac_encode_flush(AVCodecContext *avctx) void *out_ptr = dummy_out; AACENC_ERROR err; - ff_af_queue_remove(&s->afq, s->afq.frame_count, &pts, &duration); + ff_af_queue_remove(&s->afq, s->afq.frame_count, NULL); in_buf.bufs = (void **)inBuffer; in_buf.numBufs = s->metadata_mode == 0 ? 1 : 2; @@ -468,7 +467,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, int out_buffer_identifier = OUT_BITSTREAM_DATA; int out_buffer_size, out_buffer_element_size; void *out_ptr; - int ret, discard_padding; + int ret; uint8_t dummy_buf[1]; AACENC_ERROR err; @@ -528,28 +527,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, return 0; /* Get the next frame pts & duration */ - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); - - discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration); - // Check if subtraction resulted in an overflow - if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) { - av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n"); - return AVERROR(EINVAL); - } - - if (s->delay > 0 || discard_padding > 0) { - uint8_t *side_data = - av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10); - if (!side_data) - return AVERROR(ENOMEM); - if (s->delay) { - AV_WL32(side_data, FFMIN(s->delay, ff_samples_from_time_base(avctx, avpkt->duration))); - s->delay -= ff_samples_from_time_base(avctx, avpkt->duration); - s->delay = FFMAX(s->delay, 0); - } - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; avpkt->size = out_args.numOutBytes; avpkt->flags |= AV_PKT_FLAG_KEY; diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 339505ccfe..d6aa8cf62e 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -201,7 +201,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, { LAMEContext *s = avctx->priv_data; MPADecodeHeader hdr; - int len, ret, ch, discard_padding; + int len, ret, ch; int lame_result; uint32_t h; @@ -282,27 +282,9 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, memmove(s->buffer, s->buffer + len, s->buffer_index); /* Get the next frame pts/duration */ - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); - - discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration); - // Check if subtraction resulted in an overflow - if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) { - av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n"); - return AVERROR(EINVAL); - } - if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) { - uint8_t* side_data = av_packet_new_side_data(avpkt, - AV_PKT_DATA_SKIP_SAMPLES, - 10); - if (!side_data) - return AVERROR(ENOMEM); - if (!s->delay_sent) { - AV_WL32(side_data, avctx->initial_padding); - s->delay_sent = 1; - } - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; } diff --git a/libavcodec/libopencore-amr.c b/libavcodec/libopencore-amr.c index 95c7b9b137..09d201c50f 100644 --- a/libavcodec/libopencore-amr.c +++ b/libavcodec/libopencore-amr.c @@ -281,8 +281,9 @@ static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, written, s->enc_mode, avpkt->data[0]); /* Get the next frame pts/duration */ - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; avpkt->size = written; *got_packet_ptr = 1; diff --git a/libavcodec/libopusenc.c b/libavcodec/libopusenc.c index 394911ff36..9782dd9cdc 100644 --- a/libavcodec/libopusenc.c +++ b/libavcodec/libopusenc.c @@ -471,7 +471,6 @@ static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, const int sample_size = channels * bytes_per_sample; const uint8_t *audio; int ret; - int discard_padding; if (frame) { ret = ff_af_queue_add(&opus->afq, frame); @@ -517,21 +516,9 @@ static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, av_shrink_packet(avpkt, ret); - ff_af_queue_remove(&opus->afq, opus->opts.packet_size, - &avpkt->pts, &avpkt->duration); - - discard_padding = opus->opts.packet_size - ff_samples_from_time_base(avctx, avpkt->duration); - // Check if subtraction resulted in an overflow - if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) - return AVERROR(EINVAL); - if (discard_padding > 0) { - uint8_t* side_data = av_packet_new_side_data(avpkt, - AV_PKT_DATA_SKIP_SAMPLES, - 10); - if (!side_data) - return AVERROR(ENOMEM); - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&opus->afq, opus->opts.packet_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; diff --git a/libavcodec/libshine.c b/libavcodec/libshine.c index aa71383bfb..17368e5de8 100644 --- a/libavcodec/libshine.c +++ b/libavcodec/libshine.c @@ -105,8 +105,9 @@ static int libshine_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, s->buffer_index -= len; memmove(s->buffer, s->buffer + len, s->buffer_index); - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; } diff --git a/libavcodec/libspeexenc.c b/libavcodec/libspeexenc.c index 6f2d1ac7e9..32abadcef1 100644 --- a/libavcodec/libspeexenc.c +++ b/libavcodec/libspeexenc.c @@ -296,8 +296,10 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, speex_bits_reset(&s->bits); /* Get the next frame pts/duration */ - ff_af_queue_remove(&s->afq, s->frames_per_packet * avctx->frame_size, - &avpkt->pts, &avpkt->duration); + ret = ff_af_queue_remove(&s->afq, s->frames_per_packet * avctx->frame_size, + avpkt); + if (ret < 0) + return ret; avpkt->size = ret; *got_packet_ptr = 1; diff --git a/libavcodec/libvorbisenc.c b/libavcodec/libvorbisenc.c index 352a5e1297..5ae65df32c 100644 --- a/libavcodec/libvorbisenc.c +++ b/libavcodec/libvorbisenc.c @@ -415,19 +415,9 @@ static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, duration = av_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size); if (duration > 0) { - int discard_padding; - - ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration); - - discard_padding = duration - ff_samples_from_time_base(avctx, avpkt->duration); - if (discard_padding > 0) { - uint8_t *side_data = av_packet_new_side_data(avpkt, - AV_PKT_DATA_SKIP_SAMPLES, - 10); - if (!side_data) - return AVERROR(ENOMEM); - AV_WL32(side_data + 4, discard_padding); - } + ret = ff_af_queue_remove(&s->afq, duration, avpkt); + if (ret < 0) + return ret; } *got_packet_ptr = 1; diff --git a/libavcodec/mlpenc.c b/libavcodec/mlpenc.c index 4d4c33f485..ceeedb01db 100644 --- a/libavcodec/mlpenc.c +++ b/libavcodec/mlpenc.c @@ -2241,10 +2241,9 @@ input_and_return: avctx->frame_num++; if (bytes_written > 0) { - ff_af_queue_remove(&ctx->afq, - FFMIN(avctx->frame_size, ctx->afq.remaining_samples), - &avpkt->pts, - &avpkt->duration); + ret = ff_af_queue_remove(&ctx->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; av_shrink_packet(avpkt, bytes_written); diff --git a/libavcodec/nellymoserenc.c b/libavcodec/nellymoserenc.c index 6f002eb891..54b598dbb1 100644 --- a/libavcodec/nellymoserenc.c +++ b/libavcodec/nellymoserenc.c @@ -409,8 +409,9 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, encode_block(s, avpkt->data, avpkt->size); /* Get the next frame pts/duration */ - ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); + ret = ff_af_queue_remove(&s->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; return 0; diff --git a/libavcodec/opus/enc.c b/libavcodec/opus/enc.c index 74531f9142..79faade59a 100644 --- a/libavcodec/opus/enc.c +++ b/libavcodec/opus/enc.c @@ -611,7 +611,9 @@ static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, ff_opus_psy_postencode_update(&s->psyctx, s->frame); /* Remove samples from queue and skip if needed */ - ff_af_queue_remove(&s->afq, s->packet.frames*frame_size, &avpkt->pts, &avpkt->duration); + ret = ff_af_queue_remove(&s->afq, s->packet.frames*frame_size, avpkt); + if (ret < 0) + return ret; discard_padding = s->packet.frames*frame_size - ff_samples_from_time_base(avctx, avpkt->duration); if (discard_padding > 0) { diff --git a/libavcodec/ra144enc.c b/libavcodec/ra144enc.c index d38c39ce14..dbe6dcd145 100644 --- a/libavcodec/ra144enc.c +++ b/libavcodec/ra144enc.c @@ -527,8 +527,9 @@ static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block)); /* Get the next frame pts/duration */ - ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts, - &avpkt->duration); + ret = ff_af_queue_remove(&ractx->afq, avctx->frame_size, avpkt); + if (ret < 0) + return ret; *got_packet_ptr = 1; return 0; diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c index ab30dd49ed..fbb51b929d 100644 --- a/libavcodec/vorbisenc.c +++ b/libavcodec/vorbisenc.c @@ -1196,7 +1196,9 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, flush_put_bits(&pb); avpkt->size = put_bytes_output(&pb); - ff_af_queue_remove(&venc->afq, frame_size, &avpkt->pts, &avpkt->duration); + ret = ff_af_queue_remove(&venc->afq, frame_size, avpkt); + if (ret < 0) + return ret; if (frame_size > avpkt->duration) { uint8_t *side = av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10); diff --git a/tests/ref/fate/ac3-fixed-encode-2 b/tests/ref/fate/ac3-fixed-encode-2 index 468441cd7e..c01184806b 100644 --- a/tests/ref/fate/ac3-fixed-encode-2 +++ b/tests/ref/fate/ac3-fixed-encode-2 @@ -3,7 +3,7 @@ #codec_id 0: ac3 #sample_rate 0: 44100 #channel_layout_name 0: 5.1(side) -0, -256, -256, 1536, 1114, 0x32fd276c +0, -256, -256, 1536, 1114, 0x32fd276c, S=1, Skip Samples, 10, 0x00090001 0, 1280, 1280, 1536, 1116, 0x1ac63ba7 0, 2816, 2816, 1536, 1114, 0xdde82dbc 0, 4352, 4352, 1536, 1114, 0x39313179 diff --git a/tests/ref/fate/ac3-fixed-encode-3 b/tests/ref/fate/ac3-fixed-encode-3 index 041bb8c636..7171043753 100644 --- a/tests/ref/fate/ac3-fixed-encode-3 +++ b/tests/ref/fate/ac3-fixed-encode-3 @@ -3,7 +3,7 @@ #codec_id 0: ac3 #sample_rate 0: 44100 #channel_layout_name 0: 5.1 -0, -256, -256, 1536, 1114, 0x89963331 +0, -256, -256, 1536, 1114, 0x89963331, S=1, Skip Samples, 10, 0x00090001 0, 1280, 1280, 1536, 1116, 0x9eda26ff 0, 2816, 2816, 1536, 1114, 0x0734390b 0, 4352, 4352, 1536, 1114, 0xa6e33728 diff --git a/tests/ref/fate/copy-shortest1 b/tests/ref/fate/copy-shortest1 index 83a73d5263..d8ba4a59f9 100644 --- a/tests/ref/fate/copy-shortest1 +++ b/tests/ref/fate/copy-shortest1 @@ -13,7 +13,7 @@ #sample_rate 1: 44100 #channel_layout_name 1: mono #stream#, dts, pts, duration, size, hash -1, -256, -256, 1536, 416, 180f042a77b9500f9a002cafd2f670a2 +1, -256, -256, 1536, 416, 180f042a77b9500f9a002cafd2f670a2, S=1, 10, 4bce5775f7be5f45922731da9a33b4f3 0, 0, 0, 2048, 8719, bbea2a7487d61d39a0b2f2fe62a4df4a 1, 1280, 1280, 1536, 418, 77effcb2892958193be38a788328616b 0, 2048, 2048, 2048, 975, 94f30e410595452ee981d96224516504 diff --git a/tests/ref/fate/copy-shortest2 b/tests/ref/fate/copy-shortest2 index 83a73d5263..d8ba4a59f9 100644 --- a/tests/ref/fate/copy-shortest2 +++ b/tests/ref/fate/copy-shortest2 @@ -13,7 +13,7 @@ #sample_rate 1: 44100 #channel_layout_name 1: mono #stream#, dts, pts, duration, size, hash -1, -256, -256, 1536, 416, 180f042a77b9500f9a002cafd2f670a2 +1, -256, -256, 1536, 416, 180f042a77b9500f9a002cafd2f670a2, S=1, 10, 4bce5775f7be5f45922731da9a33b4f3 0, 0, 0, 2048, 8719, bbea2a7487d61d39a0b2f2fe62a4df4a 1, 1280, 1280, 1536, 418, 77effcb2892958193be38a788328616b 0, 2048, 2048, 2048, 975, 94f30e410595452ee981d96224516504 diff --git a/tests/ref/fate/ffmpeg-filter_complex_audio b/tests/ref/fate/ffmpeg-filter_complex_audio index 5b42ab1b8a..c178e0bd94 100644 --- a/tests/ref/fate/ffmpeg-filter_complex_audio +++ b/tests/ref/fate/ffmpeg-filter_complex_audio @@ -3,7 +3,7 @@ #codec_id 0: ac3 #sample_rate 0: 44100 #channel_layout_name 0: mono -0, -256, -256, 1536, 416, 0x3001fb2d +0, -256, -256, 1536, 416, 0x3001fb2d, S=1, Skip Samples, 10, 0x00090001 0, 1280, 1280, 1536, 418, 0xba72fc16 0, 2816, 2816, 1536, 418, 0xba72fc16 0, 4352, 4352, 259, 418, 0xba72fc16, S=1, Skip Samples, 10, 0x06020101 diff --git a/tests/ref/fate/opus-enc-silence b/tests/ref/fate/opus-enc-silence index dd8e02e02d..8d3e71e1ca 100644 --- a/tests/ref/fate/opus-enc-silence +++ b/tests/ref/fate/opus-enc-silence @@ -4,5 +4,5 @@ #codec_id 0: opus #sample_rate 0: 48000 #channel_layout_name 0: stereo -0, -120, -120, 960, 1, 0x00fc00fc +0, -120, -120, 960, 1, 0x00fc00fc, S=1, Skip Samples, 10, 0x04b00078 0, 840, 840, 168, 1, 0x00fc00fc, S=1, Skip Samples, 10, 0x009f001b diff --git a/tests/ref/fate/shortest b/tests/ref/fate/shortest index 68d1038b71..8343db4872 100644 --- a/tests/ref/fate/shortest +++ b/tests/ref/fate/shortest @@ -8,7 +8,7 @@ #codec_id 1: ac3 #sample_rate 1: 44100 #channel_layout_name 1: mono -1, -256, -256, 1536, 416, 0xcedecce4 +1, -256, -256, 1536, 416, 0xcedecce4, S=1, Skip Samples, 10, 0x00090001 0, 0, 0, 1, 27867, 0x1426a0d6, S=1, Quality stats, 8, 0x050000a1 1, 1280, 1280, 1536, 418, 0x4ebabf82 0, 1, 1, 1, 9806, 0xbebc2826, F=0x0, S=1, Quality stats, 8, 0x050400a2 _______________________________________________ ffmpeg-cvslog mailing list -- [email protected] To unsubscribe send an email to [email protected]
