| Message ID | 1474406986-31418-1-git-send-email-isasi@google.com |
|---|---|
| State | Superseded |
| Headers | show |
friendly ping On Tue, Sep 20, 2016 at 2:29 PM, Sasi Inguva <isasi@google.com> wrote: > Signed-off-by: Sasi Inguva <isasi@google.com> > --- > libavcodec/utils.c | 15 +++--- > libavformat/mov.c | 81 > ++++++++++++++++++++++++---- > tests/ref/fate/gaplessenc-itunes-to-ipod-aac | 2 +- > tests/ref/fate/gaplessenc-pcm-to-mov-aac | 2 +- > 4 files changed, 78 insertions(+), 22 deletions(-) > > diff --git a/libavcodec/utils.c b/libavcodec/utils.c > index b0345b6..e18476c 100644 > --- a/libavcodec/utils.c > +++ b/libavcodec/utils.c > @@ -2320,7 +2320,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > uint32_t discard_padding = 0; > uint8_t skip_reason = 0; > uint8_t discard_reason = 0; > - int demuxer_skip_samples = 0; > // copy to ensure we do not change avpkt > AVPacket tmp = *avpkt; > int did_split = av_packet_split_side_data(&tmp); > @@ -2328,7 +2327,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > if (ret < 0) > goto fail; > > - demuxer_skip_samples = avctx->internal->skip_samples; > avctx->internal->pkt = &tmp; > if (HAVE_THREADS && avctx->active_thread_type & FF_THREAD_FRAME) > ret = ff_thread_decode_frame(avctx, frame, got_frame_ptr, > &tmp); > @@ -2353,13 +2351,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > frame->sample_rate = avctx->sample_rate; > } > > - > - if (frame->flags & AV_FRAME_FLAG_DISCARD) { > - // If using discard frame flag, ignore skip_samples set by > the decoder. > - avctx->internal->skip_samples = demuxer_skip_samples; > - *got_frame_ptr = 0; > - } > - > side= av_packet_get_side_data(avctx->internal->pkt, > AV_PKT_DATA_SKIP_SAMPLES, &side_size); > if(side && side_size>=10) { > avctx->internal->skip_samples = AV_RL32(side); > @@ -2369,6 +2360,12 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > skip_reason = AV_RL8(side + 8); > discard_reason = AV_RL8(side + 9); > } > + > + if ((frame->flags & AV_FRAME_FLAG_DISCARD) && *got_frame_ptr) { > + avctx->internal->skip_samples -= frame->nb_samples; > + *got_frame_ptr = 0; > + } > + > if (avctx->internal->skip_samples > 0 && *got_frame_ptr && > !(avctx->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) { > if(frame->nb_samples <= avctx->internal->skip_samples){ > diff --git a/libavformat/mov.c b/libavformat/mov.c > index b84d9c0..bb86780 100644 > --- a/libavformat/mov.c > +++ b/libavformat/mov.c > @@ -2856,6 +2856,21 @@ static int64_t add_index_entry(AVStream *st, > int64_t pos, int64_t timestamp, > } > > /** > + * Rewrite timestamps of index entries in the range [end_index - > frame_duration_buffer_size, end_index) > + * by subtracting end_ts successively by the amounts given in > frame_duration_buffer. > + */ > +static void fix_index_entry_timestamps(AVStream* st, int end_index, > int64_t end_ts, > + int64_t* frame_duration_buffer, > + int frame_duration_buffer_size) { > + int i = 0; > + av_assert0(end_index >= 0 && end_index <= st->nb_index_entries); > + for (i = 0; i < frame_duration_buffer_size; i++) { > + end_ts -= frame_duration_buffer[frame_duration_buffer_size - 1 - > i]; > + st->index_entries[end_index - 1 - i].timestamp = end_ts; > + } > +} > + > +/** > * Append a new ctts entry to ctts_data. > * Returns the new ctts_count if successful, else returns -1. > */ > @@ -2919,7 +2934,10 @@ static void mov_fix_index(MOVContext *mov, AVStream > *st) > int64_t edit_list_media_time_dts = 0; > int64_t edit_list_start_encountered = 0; > int64_t search_timestamp = 0; > - > + int64_t* frame_duration_buffer = NULL; > + int num_discarded_begin = 0; > + int first_non_zero_audio_edit = -1; > + int packet_skip_samples = 0; > > if (!msc->elst_data || msc->elst_count <= 0) { > return; > @@ -2955,6 +2973,7 @@ static void mov_fix_index(MOVContext *mov, AVStream > *st) > edit_list_index++; > edit_list_dts_counter = edit_list_dts_entry_end; > edit_list_dts_entry_end += edit_list_duration; > + num_discarded_begin = 0; > if (edit_list_media_time == -1) { > continue; > } > @@ -2962,7 +2981,14 @@ static void mov_fix_index(MOVContext *mov, AVStream > *st) > // If we encounter a non-negative edit list reset the > skip_samples/start_pad fields and set them > // according to the edit list below. > if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { > - st->skip_samples = msc->start_pad = 0; > + if (first_non_zero_audio_edit < 0) { > + first_non_zero_audio_edit = 1; > + } else { > + first_non_zero_audio_edit = 0; > + } > + > + if (first_non_zero_audio_edit > 0) > + st->skip_samples = msc->start_pad = 0; > } > > //find closest previous key frame > @@ -3041,24 +3067,57 @@ static void mov_fix_index(MOVContext *mov, > AVStream *st) > } > > if (curr_cts < edit_list_media_time || curr_cts >= > (edit_list_duration + edit_list_media_time)) { > - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > curr_cts < edit_list_media_time && > - curr_cts + frame_duration > edit_list_media_time && > - st->skip_samples == 0 && msc->start_pad == 0) { > - st->skip_samples = msc->start_pad = > edit_list_media_time - curr_cts; > - > - // Shift the index entry timestamp by skip_samples to > be correct. > - edit_list_dts_counter -= st->skip_samples; > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > st->codecpar->codec_id != AV_CODEC_ID_VORBIS && > + curr_cts < edit_list_media_time && curr_cts + > frame_duration > edit_list_media_time && > + first_non_zero_audio_edit > 0) { > + packet_skip_samples = edit_list_media_time - > curr_cts; > + st->skip_samples += packet_skip_samples; > + > + // Shift the index entry timestamp by > packet_skip_samples to be correct. > + edit_list_dts_counter -= packet_skip_samples; > if (edit_list_start_encountered == 0) { > - edit_list_start_encountered = 1; > + edit_list_start_encountered = 1; > + // Make timestamps strictly monotonically > increasing for audio, by rewriting timestamps for > + // discarded packets. > + if (frame_duration_buffer) { > + fix_index_entry_timestamps(st, > st->nb_index_entries, edit_list_dts_counter, > + > frame_duration_buffer, num_discarded_begin); > + av_freep(&frame_duration_buffer); > + } > } > > - av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples > from curr_cts: %"PRId64"\n", st->skip_samples, curr_cts); > + av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples > from curr_cts: %"PRId64"\n", packet_skip_samples, curr_cts); > } else { > flags |= AVINDEX_DISCARD_FRAME; > av_log(mov->fc, AV_LOG_DEBUG, "drop a frame at > curr_cts: %"PRId64" @ %"PRId64"\n", curr_cts, index); > + > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > edit_list_start_encountered == 0) { > + num_discarded_begin++; > + frame_duration_buffer = av_realloc(frame_duration_ > buffer, > + > num_discarded_begin * sizeof(int64_t)); > + if (!frame_duration_buffer) { > + av_log(mov->fc, AV_LOG_ERROR, "Cannot > reallocate frame duration buffer\n"); > + break; > + } > + frame_duration_buffer[num_discarded_begin - 1] = > frame_duration; > + > + // Increment skip_samples for the first non-zero > audio edit list > + if (first_non_zero_audio_edit > 0 && > st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { > + st->skip_samples += frame_duration; > + msc->start_pad = st->skip_samples; > + } > + } > } > } else if (edit_list_start_encountered == 0) { > edit_list_start_encountered = 1; > + // Make timestamps strictly monotonically increasing for > audio, by rewriting timestamps for > + // discarded packets. > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > frame_duration_buffer) { > + fix_index_entry_timestamps(st, st->nb_index_entries, > edit_list_dts_counter, > + frame_duration_buffer, > num_discarded_begin); > + av_freep(&frame_duration_buffer); > + } > + > } > > if (add_index_entry(st, current->pos, edit_list_dts_counter, > current->size, > diff --git a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > index 043c085..789681f 100644 > --- a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > +++ b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > @@ -7,7 +7,7 @@ duration_ts=103326 > start_time=0.000000 > duration=2.367000 > [/FORMAT] > -packet|pts=0|dts=0|duration=N/A > +packet|pts=-1024|dts=-1024|duration=1024 > packet|pts=0|dts=0|duration=1024 > packet|pts=1024|dts=1024|duration=1024 > packet|pts=2048|dts=2048|duration=1024 > diff --git a/tests/ref/fate/gaplessenc-pcm-to-mov-aac > b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > index 8b7e3f6..8702611 100644 > --- a/tests/ref/fate/gaplessenc-pcm-to-mov-aac > +++ b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > @@ -7,7 +7,7 @@ duration_ts=529200 > start_time=0.000000 > duration=12.024000 > [/FORMAT] > -packet|pts=0|dts=0|duration=N/A > +packet|pts=-1024|dts=-1024|duration=1024 > packet|pts=0|dts=0|duration=1024 > packet|pts=1024|dts=1024|duration=1024 > packet|pts=2048|dts=2048|duration=1024 > -- > 2.8.0.rc3.226.g39d4020 > >
On Tue, 20 Sep 2016 14:29:46 -0700 Sasi Inguva <isasi-at-google.com@ffmpeg.org> wrote: > Signed-off-by: Sasi Inguva <isasi@google.com> > --- > libavcodec/utils.c | 15 +++--- > libavformat/mov.c | 81 ++++++++++++++++++++++++---- > tests/ref/fate/gaplessenc-itunes-to-ipod-aac | 2 +- > tests/ref/fate/gaplessenc-pcm-to-mov-aac | 2 +- > 4 files changed, 78 insertions(+), 22 deletions(-) > > diff --git a/libavcodec/utils.c b/libavcodec/utils.c > index b0345b6..e18476c 100644 > --- a/libavcodec/utils.c > +++ b/libavcodec/utils.c > @@ -2320,7 +2320,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, > uint32_t discard_padding = 0; > uint8_t skip_reason = 0; > uint8_t discard_reason = 0; > - int demuxer_skip_samples = 0; > // copy to ensure we do not change avpkt > AVPacket tmp = *avpkt; > int did_split = av_packet_split_side_data(&tmp); > @@ -2328,7 +2327,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, > if (ret < 0) > goto fail; > > - demuxer_skip_samples = avctx->internal->skip_samples; > avctx->internal->pkt = &tmp; > if (HAVE_THREADS && avctx->active_thread_type & FF_THREAD_FRAME) > ret = ff_thread_decode_frame(avctx, frame, got_frame_ptr, &tmp); > @@ -2353,13 +2351,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, > frame->sample_rate = avctx->sample_rate; > } > > - > - if (frame->flags & AV_FRAME_FLAG_DISCARD) { > - // If using discard frame flag, ignore skip_samples set by the decoder. > - avctx->internal->skip_samples = demuxer_skip_samples; > - *got_frame_ptr = 0; > - } > - > side= av_packet_get_side_data(avctx->internal->pkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size); > if(side && side_size>=10) { > avctx->internal->skip_samples = AV_RL32(side); > @@ -2369,6 +2360,12 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, > skip_reason = AV_RL8(side + 8); > discard_reason = AV_RL8(side + 9); > } > + > + if ((frame->flags & AV_FRAME_FLAG_DISCARD) && *got_frame_ptr) { > + avctx->internal->skip_samples -= frame->nb_samples; > + *got_frame_ptr = 0; > + } > + > if (avctx->internal->skip_samples > 0 && *got_frame_ptr && > !(avctx->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) { > if(frame->nb_samples <= avctx->internal->skip_samples){ > diff --git a/libavformat/mov.c b/libavformat/mov.c > index b84d9c0..bb86780 100644 > --- a/libavformat/mov.c > +++ b/libavformat/mov.c > @@ -2856,6 +2856,21 @@ static int64_t add_index_entry(AVStream *st, int64_t pos, int64_t timestamp, > } > > /** > + * Rewrite timestamps of index entries in the range [end_index - frame_duration_buffer_size, end_index) > + * by subtracting end_ts successively by the amounts given in frame_duration_buffer. > + */ > +static void fix_index_entry_timestamps(AVStream* st, int end_index, int64_t end_ts, > + int64_t* frame_duration_buffer, > + int frame_duration_buffer_size) { > + int i = 0; > + av_assert0(end_index >= 0 && end_index <= st->nb_index_entries); > + for (i = 0; i < frame_duration_buffer_size; i++) { > + end_ts -= frame_duration_buffer[frame_duration_buffer_size - 1 - i]; > + st->index_entries[end_index - 1 - i].timestamp = end_ts; > + } > +} > + > +/** > * Append a new ctts entry to ctts_data. > * Returns the new ctts_count if successful, else returns -1. > */ > @@ -2919,7 +2934,10 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) > int64_t edit_list_media_time_dts = 0; > int64_t edit_list_start_encountered = 0; > int64_t search_timestamp = 0; > - > + int64_t* frame_duration_buffer = NULL; > + int num_discarded_begin = 0; > + int first_non_zero_audio_edit = -1; > + int packet_skip_samples = 0; > > if (!msc->elst_data || msc->elst_count <= 0) { > return; > @@ -2955,6 +2973,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) > edit_list_index++; > edit_list_dts_counter = edit_list_dts_entry_end; > edit_list_dts_entry_end += edit_list_duration; > + num_discarded_begin = 0; > if (edit_list_media_time == -1) { > continue; > } > @@ -2962,7 +2981,14 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) > // If we encounter a non-negative edit list reset the skip_samples/start_pad fields and set them > // according to the edit list below. > if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { > - st->skip_samples = msc->start_pad = 0; > + if (first_non_zero_audio_edit < 0) { > + first_non_zero_audio_edit = 1; > + } else { > + first_non_zero_audio_edit = 0; > + } > + > + if (first_non_zero_audio_edit > 0) > + st->skip_samples = msc->start_pad = 0; > } > > //find closest previous key frame > @@ -3041,24 +3067,57 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) > } > > if (curr_cts < edit_list_media_time || curr_cts >= (edit_list_duration + edit_list_media_time)) { > - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && curr_cts < edit_list_media_time && > - curr_cts + frame_duration > edit_list_media_time && > - st->skip_samples == 0 && msc->start_pad == 0) { > - st->skip_samples = msc->start_pad = edit_list_media_time - curr_cts; > - > - // Shift the index entry timestamp by skip_samples to be correct. > - edit_list_dts_counter -= st->skip_samples; > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->codec_id != AV_CODEC_ID_VORBIS && > + curr_cts < edit_list_media_time && curr_cts + frame_duration > edit_list_media_time && > + first_non_zero_audio_edit > 0) { > + packet_skip_samples = edit_list_media_time - curr_cts; > + st->skip_samples += packet_skip_samples; > + > + // Shift the index entry timestamp by packet_skip_samples to be correct. > + edit_list_dts_counter -= packet_skip_samples; > if (edit_list_start_encountered == 0) { > - edit_list_start_encountered = 1; > + edit_list_start_encountered = 1; > + // Make timestamps strictly monotonically increasing for audio, by rewriting timestamps for > + // discarded packets. > + if (frame_duration_buffer) { > + fix_index_entry_timestamps(st, st->nb_index_entries, edit_list_dts_counter, > + frame_duration_buffer, num_discarded_begin); > + av_freep(&frame_duration_buffer); > + } > } > > - av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples from curr_cts: %"PRId64"\n", st->skip_samples, curr_cts); > + av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples from curr_cts: %"PRId64"\n", packet_skip_samples, curr_cts); > } else { > flags |= AVINDEX_DISCARD_FRAME; > av_log(mov->fc, AV_LOG_DEBUG, "drop a frame at curr_cts: %"PRId64" @ %"PRId64"\n", curr_cts, index); > + > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && edit_list_start_encountered == 0) { > + num_discarded_begin++; > + frame_duration_buffer = av_realloc(frame_duration_buffer, > + num_discarded_begin * sizeof(int64_t)); > + if (!frame_duration_buffer) { > + av_log(mov->fc, AV_LOG_ERROR, "Cannot reallocate frame duration buffer\n"); > + break; > + } > + frame_duration_buffer[num_discarded_begin - 1] = frame_duration; > + > + // Increment skip_samples for the first non-zero audio edit list > + if (first_non_zero_audio_edit > 0 && st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { > + st->skip_samples += frame_duration; > + msc->start_pad = st->skip_samples; > + } > + } > } > } else if (edit_list_start_encountered == 0) { > edit_list_start_encountered = 1; > + // Make timestamps strictly monotonically increasing for audio, by rewriting timestamps for > + // discarded packets. > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && frame_duration_buffer) { > + fix_index_entry_timestamps(st, st->nb_index_entries, edit_list_dts_counter, > + frame_duration_buffer, num_discarded_begin); > + av_freep(&frame_duration_buffer); > + } > + > } > > if (add_index_entry(st, current->pos, edit_list_dts_counter, current->size, > diff --git a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > index 043c085..789681f 100644 > --- a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > +++ b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > @@ -7,7 +7,7 @@ duration_ts=103326 > start_time=0.000000 > duration=2.367000 > [/FORMAT] > -packet|pts=0|dts=0|duration=N/A > +packet|pts=-1024|dts=-1024|duration=1024 > packet|pts=0|dts=0|duration=1024 > packet|pts=1024|dts=1024|duration=1024 > packet|pts=2048|dts=2048|duration=1024 > diff --git a/tests/ref/fate/gaplessenc-pcm-to-mov-aac b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > index 8b7e3f6..8702611 100644 > --- a/tests/ref/fate/gaplessenc-pcm-to-mov-aac > +++ b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > @@ -7,7 +7,7 @@ duration_ts=529200 > start_time=0.000000 > duration=12.024000 > [/FORMAT] > -packet|pts=0|dts=0|duration=N/A > +packet|pts=-1024|dts=-1024|duration=1024 > packet|pts=0|dts=0|duration=1024 > packet|pts=1024|dts=1024|duration=1024 > packet|pts=2048|dts=2048|duration=1024 This is a complex patch, and builds upon new code that isn't quite known to me, the result looks like an improvement to me. Does it work with the "skip_manual" libavcodec option? I also think the fate tests should be updated to include the packet flags, since they are just as important as the timestamps.
On Thu, Sep 22, 2016 at 5:49 AM, wm4 <nfxjfg@googlemail.com> wrote: > On Tue, 20 Sep 2016 14:29:46 -0700 > Sasi Inguva <isasi-at-google.com@ffmpeg.org> wrote: > > > Signed-off-by: Sasi Inguva <isasi@google.com> > > --- > > libavcodec/utils.c | 15 +++--- > > libavformat/mov.c | 81 > ++++++++++++++++++++++++---- > > tests/ref/fate/gaplessenc-itunes-to-ipod-aac | 2 +- > > tests/ref/fate/gaplessenc-pcm-to-mov-aac | 2 +- > > 4 files changed, 78 insertions(+), 22 deletions(-) > > > > diff --git a/libavcodec/utils.c b/libavcodec/utils.c > > index b0345b6..e18476c 100644 > > --- a/libavcodec/utils.c > > +++ b/libavcodec/utils.c > > @@ -2320,7 +2320,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > > uint32_t discard_padding = 0; > > uint8_t skip_reason = 0; > > uint8_t discard_reason = 0; > > - int demuxer_skip_samples = 0; > > // copy to ensure we do not change avpkt > > AVPacket tmp = *avpkt; > > int did_split = av_packet_split_side_data(&tmp); > > @@ -2328,7 +2327,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > > if (ret < 0) > > goto fail; > > > > - demuxer_skip_samples = avctx->internal->skip_samples; > > avctx->internal->pkt = &tmp; > > if (HAVE_THREADS && avctx->active_thread_type & FF_THREAD_FRAME) > > ret = ff_thread_decode_frame(avctx, frame, got_frame_ptr, > &tmp); > > @@ -2353,13 +2351,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > > frame->sample_rate = avctx->sample_rate; > > } > > > > - > > - if (frame->flags & AV_FRAME_FLAG_DISCARD) { > > - // If using discard frame flag, ignore skip_samples set by > the decoder. > > - avctx->internal->skip_samples = demuxer_skip_samples; > > - *got_frame_ptr = 0; > > - } > > - > > side= av_packet_get_side_data(avctx->internal->pkt, > AV_PKT_DATA_SKIP_SAMPLES, &side_size); > > if(side && side_size>=10) { > > avctx->internal->skip_samples = AV_RL32(side); > > @@ -2369,6 +2360,12 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext > *avctx, > > skip_reason = AV_RL8(side + 8); > > discard_reason = AV_RL8(side + 9); > > } > > + > > + if ((frame->flags & AV_FRAME_FLAG_DISCARD) && *got_frame_ptr) { > > + avctx->internal->skip_samples -= frame->nb_samples; > > + *got_frame_ptr = 0; > > + } > > + > > if (avctx->internal->skip_samples > 0 && *got_frame_ptr && > > !(avctx->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) { > > if(frame->nb_samples <= avctx->internal->skip_samples){ > > diff --git a/libavformat/mov.c b/libavformat/mov.c > > index b84d9c0..bb86780 100644 > > --- a/libavformat/mov.c > > +++ b/libavformat/mov.c > > @@ -2856,6 +2856,21 @@ static int64_t add_index_entry(AVStream *st, > int64_t pos, int64_t timestamp, > > } > > > > /** > > + * Rewrite timestamps of index entries in the range [end_index - > frame_duration_buffer_size, end_index) > > + * by subtracting end_ts successively by the amounts given in > frame_duration_buffer. > > + */ > > +static void fix_index_entry_timestamps(AVStream* st, int end_index, > int64_t end_ts, > > + int64_t* frame_duration_buffer, > > + int frame_duration_buffer_size) { > > + int i = 0; > > + av_assert0(end_index >= 0 && end_index <= st->nb_index_entries); > > + for (i = 0; i < frame_duration_buffer_size; i++) { > > + end_ts -= frame_duration_buffer[frame_duration_buffer_size - 1 > - i]; > > + st->index_entries[end_index - 1 - i].timestamp = end_ts; > > + } > > +} > > + > > +/** > > * Append a new ctts entry to ctts_data. > > * Returns the new ctts_count if successful, else returns -1. > > */ > > @@ -2919,7 +2934,10 @@ static void mov_fix_index(MOVContext *mov, > AVStream *st) > > int64_t edit_list_media_time_dts = 0; > > int64_t edit_list_start_encountered = 0; > > int64_t search_timestamp = 0; > > - > > + int64_t* frame_duration_buffer = NULL; > > + int num_discarded_begin = 0; > > + int first_non_zero_audio_edit = -1; > > + int packet_skip_samples = 0; > > > > if (!msc->elst_data || msc->elst_count <= 0) { > > return; > > @@ -2955,6 +2973,7 @@ static void mov_fix_index(MOVContext *mov, > AVStream *st) > > edit_list_index++; > > edit_list_dts_counter = edit_list_dts_entry_end; > > edit_list_dts_entry_end += edit_list_duration; > > + num_discarded_begin = 0; > > if (edit_list_media_time == -1) { > > continue; > > } > > @@ -2962,7 +2981,14 @@ static void mov_fix_index(MOVContext *mov, > AVStream *st) > > // If we encounter a non-negative edit list reset the > skip_samples/start_pad fields and set them > > // according to the edit list below. > > if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { > > - st->skip_samples = msc->start_pad = 0; > > + if (first_non_zero_audio_edit < 0) { > > + first_non_zero_audio_edit = 1; > > + } else { > > + first_non_zero_audio_edit = 0; > > + } > > + > > + if (first_non_zero_audio_edit > 0) > > + st->skip_samples = msc->start_pad = 0; > > } > > > > //find closest previous key frame > > @@ -3041,24 +3067,57 @@ static void mov_fix_index(MOVContext *mov, > AVStream *st) > > } > > > > if (curr_cts < edit_list_media_time || curr_cts >= > (edit_list_duration + edit_list_media_time)) { > > - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > curr_cts < edit_list_media_time && > > - curr_cts + frame_duration > edit_list_media_time && > > - st->skip_samples == 0 && msc->start_pad == 0) { > > - st->skip_samples = msc->start_pad = > edit_list_media_time - curr_cts; > > - > > - // Shift the index entry timestamp by skip_samples > to be correct. > > - edit_list_dts_counter -= st->skip_samples; > > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > st->codecpar->codec_id != AV_CODEC_ID_VORBIS && > > + curr_cts < edit_list_media_time && curr_cts + > frame_duration > edit_list_media_time && > > + first_non_zero_audio_edit > 0) { > > + packet_skip_samples = edit_list_media_time - > curr_cts; > > + st->skip_samples += packet_skip_samples; > > + > > + // Shift the index entry timestamp by > packet_skip_samples to be correct. > > + edit_list_dts_counter -= packet_skip_samples; > > if (edit_list_start_encountered == 0) { > > - edit_list_start_encountered = 1; > > + edit_list_start_encountered = 1; > > + // Make timestamps strictly monotonically > increasing for audio, by rewriting timestamps for > > + // discarded packets. > > + if (frame_duration_buffer) { > > + fix_index_entry_timestamps(st, > st->nb_index_entries, edit_list_dts_counter, > > + > frame_duration_buffer, num_discarded_begin); > > + av_freep(&frame_duration_buffer); > > + } > > } > > > > - av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio > samples from curr_cts: %"PRId64"\n", st->skip_samples, curr_cts); > > + av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio > samples from curr_cts: %"PRId64"\n", packet_skip_samples, curr_cts); > > } else { > > flags |= AVINDEX_DISCARD_FRAME; > > av_log(mov->fc, AV_LOG_DEBUG, "drop a frame at > curr_cts: %"PRId64" @ %"PRId64"\n", curr_cts, index); > > + > > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO > && edit_list_start_encountered == 0) { > > + num_discarded_begin++; > > + frame_duration_buffer = > av_realloc(frame_duration_buffer, > > + > num_discarded_begin * sizeof(int64_t)); > > + if (!frame_duration_buffer) { > > + av_log(mov->fc, AV_LOG_ERROR, "Cannot > reallocate frame duration buffer\n"); > > + break; > > + } > > + frame_duration_buffer[num_discarded_begin - 1] > = frame_duration; > > + > > + // Increment skip_samples for the first > non-zero audio edit list > > + if (first_non_zero_audio_edit > 0 && > st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { > > + st->skip_samples += frame_duration; > > + msc->start_pad = st->skip_samples; > > + } > > + } > > } > > } else if (edit_list_start_encountered == 0) { > > edit_list_start_encountered = 1; > > + // Make timestamps strictly monotonically increasing > for audio, by rewriting timestamps for > > + // discarded packets. > > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && > frame_duration_buffer) { > > + fix_index_entry_timestamps(st, > st->nb_index_entries, edit_list_dts_counter, > > + frame_duration_buffer, > num_discarded_begin); > > + av_freep(&frame_duration_buffer); > > + } > > + > > } > > > > if (add_index_entry(st, current->pos, > edit_list_dts_counter, current->size, > > diff --git a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > > index 043c085..789681f 100644 > > --- a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > > +++ b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac > > @@ -7,7 +7,7 @@ duration_ts=103326 > > start_time=0.000000 > > duration=2.367000 > > [/FORMAT] > > -packet|pts=0|dts=0|duration=N/A > > +packet|pts=-1024|dts=-1024|duration=1024 > > packet|pts=0|dts=0|duration=1024 > > packet|pts=1024|dts=1024|duration=1024 > > packet|pts=2048|dts=2048|duration=1024 > > diff --git a/tests/ref/fate/gaplessenc-pcm-to-mov-aac > b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > > index 8b7e3f6..8702611 100644 > > --- a/tests/ref/fate/gaplessenc-pcm-to-mov-aac > > +++ b/tests/ref/fate/gaplessenc-pcm-to-mov-aac > > @@ -7,7 +7,7 @@ duration_ts=529200 > > start_time=0.000000 > > duration=12.024000 > > [/FORMAT] > > -packet|pts=0|dts=0|duration=N/A > > +packet|pts=-1024|dts=-1024|duration=1024 > > packet|pts=0|dts=0|duration=1024 > > packet|pts=1024|dts=1024|duration=1024 > > packet|pts=2048|dts=2048|duration=1024 > > This is a complex patch, and builds upon new code that isn't quite > known to me, the result looks like an improvement to me. > > Does it work with the "skip_manual" libavcodec option? > > Nice catch. sent the patch again to make it work with skip_manual I also think the fate tests should be updated to include the packet > flags, since they are just as important as the timestamps. > I had another patch making ffprobe show the DISCARD flag. If / once that patch is applied it will be easier to modify the fate test for gapless to show the discard flag. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >
I have updated the patch with flag values for fate tests. On Thu, Sep 22, 2016 at 11:38 AM, Sasi Inguva <isasi@google.com> wrote: > > On Thu, Sep 22, 2016 at 5:49 AM, wm4 <nfxjfg@googlemail.com> wrote: > >> On Tue, 20 Sep 2016 14:29:46 -0700 >> Sasi Inguva <isasi-at-google.com@ffmpeg.org> wrote: >> >> > Signed-off-by: Sasi Inguva <isasi@google.com> >> > --- >> > libavcodec/utils.c | 15 +++--- >> > libavformat/mov.c | 81 >> ++++++++++++++++++++++++---- >> > tests/ref/fate/gaplessenc-itunes-to-ipod-aac | 2 +- >> > tests/ref/fate/gaplessenc-pcm-to-mov-aac | 2 +- >> > 4 files changed, 78 insertions(+), 22 deletions(-) >> > >> > diff --git a/libavcodec/utils.c b/libavcodec/utils.c >> > index b0345b6..e18476c 100644 >> > --- a/libavcodec/utils.c >> > +++ b/libavcodec/utils.c >> > @@ -2320,7 +2320,6 @@ int attribute_align_arg >> avcodec_decode_audio4(AVCodecContext *avctx, >> > uint32_t discard_padding = 0; >> > uint8_t skip_reason = 0; >> > uint8_t discard_reason = 0; >> > - int demuxer_skip_samples = 0; >> > // copy to ensure we do not change avpkt >> > AVPacket tmp = *avpkt; >> > int did_split = av_packet_split_side_data(&tmp); >> > @@ -2328,7 +2327,6 @@ int attribute_align_arg >> avcodec_decode_audio4(AVCodecContext *avctx, >> > if (ret < 0) >> > goto fail; >> > >> > - demuxer_skip_samples = avctx->internal->skip_samples; >> > avctx->internal->pkt = &tmp; >> > if (HAVE_THREADS && avctx->active_thread_type & >> FF_THREAD_FRAME) >> > ret = ff_thread_decode_frame(avctx, frame, got_frame_ptr, >> &tmp); >> > @@ -2353,13 +2351,6 @@ int attribute_align_arg >> avcodec_decode_audio4(AVCodecContext *avctx, >> > frame->sample_rate = avctx->sample_rate; >> > } >> > >> > - >> > - if (frame->flags & AV_FRAME_FLAG_DISCARD) { >> > - // If using discard frame flag, ignore skip_samples set by >> the decoder. >> > - avctx->internal->skip_samples = demuxer_skip_samples; >> > - *got_frame_ptr = 0; >> > - } >> > - >> > side= av_packet_get_side_data(avctx->internal->pkt, >> AV_PKT_DATA_SKIP_SAMPLES, &side_size); >> > if(side && side_size>=10) { >> > avctx->internal->skip_samples = AV_RL32(side); >> > @@ -2369,6 +2360,12 @@ int attribute_align_arg >> avcodec_decode_audio4(AVCodecContext *avctx, >> > skip_reason = AV_RL8(side + 8); >> > discard_reason = AV_RL8(side + 9); >> > } >> > + >> > + if ((frame->flags & AV_FRAME_FLAG_DISCARD) && *got_frame_ptr) { >> > + avctx->internal->skip_samples -= frame->nb_samples; >> > + *got_frame_ptr = 0; >> > + } >> > + >> > if (avctx->internal->skip_samples > 0 && *got_frame_ptr && >> > !(avctx->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) { >> > if(frame->nb_samples <= avctx->internal->skip_samples){ >> > diff --git a/libavformat/mov.c b/libavformat/mov.c >> > index b84d9c0..bb86780 100644 >> > --- a/libavformat/mov.c >> > +++ b/libavformat/mov.c >> > @@ -2856,6 +2856,21 @@ static int64_t add_index_entry(AVStream *st, >> int64_t pos, int64_t timestamp, >> > } >> > >> > /** >> > + * Rewrite timestamps of index entries in the range [end_index - >> frame_duration_buffer_size, end_index) >> > + * by subtracting end_ts successively by the amounts given in >> frame_duration_buffer. >> > + */ >> > +static void fix_index_entry_timestamps(AVStream* st, int end_index, >> int64_t end_ts, >> > + int64_t* frame_duration_buffer, >> > + int frame_duration_buffer_size) >> { >> > + int i = 0; >> > + av_assert0(end_index >= 0 && end_index <= st->nb_index_entries); >> > + for (i = 0; i < frame_duration_buffer_size; i++) { >> > + end_ts -= frame_duration_buffer[frame_duration_buffer_size - >> 1 - i]; >> > + st->index_entries[end_index - 1 - i].timestamp = end_ts; >> > + } >> > +} >> > + >> > +/** >> > * Append a new ctts entry to ctts_data. >> > * Returns the new ctts_count if successful, else returns -1. >> > */ >> > @@ -2919,7 +2934,10 @@ static void mov_fix_index(MOVContext *mov, >> AVStream *st) >> > int64_t edit_list_media_time_dts = 0; >> > int64_t edit_list_start_encountered = 0; >> > int64_t search_timestamp = 0; >> > - >> > + int64_t* frame_duration_buffer = NULL; >> > + int num_discarded_begin = 0; >> > + int first_non_zero_audio_edit = -1; >> > + int packet_skip_samples = 0; >> > >> > if (!msc->elst_data || msc->elst_count <= 0) { >> > return; >> > @@ -2955,6 +2973,7 @@ static void mov_fix_index(MOVContext *mov, >> AVStream *st) >> > edit_list_index++; >> > edit_list_dts_counter = edit_list_dts_entry_end; >> > edit_list_dts_entry_end += edit_list_duration; >> > + num_discarded_begin = 0; >> > if (edit_list_media_time == -1) { >> > continue; >> > } >> > @@ -2962,7 +2981,14 @@ static void mov_fix_index(MOVContext *mov, >> AVStream *st) >> > // If we encounter a non-negative edit list reset the >> skip_samples/start_pad fields and set them >> > // according to the edit list below. >> > if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { >> > - st->skip_samples = msc->start_pad = 0; >> > + if (first_non_zero_audio_edit < 0) { >> > + first_non_zero_audio_edit = 1; >> > + } else { >> > + first_non_zero_audio_edit = 0; >> > + } >> > + >> > + if (first_non_zero_audio_edit > 0) >> > + st->skip_samples = msc->start_pad = 0; >> > } >> > >> > //find closest previous key frame >> > @@ -3041,24 +3067,57 @@ static void mov_fix_index(MOVContext *mov, >> AVStream *st) >> > } >> > >> > if (curr_cts < edit_list_media_time || curr_cts >= >> (edit_list_duration + edit_list_media_time)) { >> > - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && >> curr_cts < edit_list_media_time && >> > - curr_cts + frame_duration > edit_list_media_time && >> > - st->skip_samples == 0 && msc->start_pad == 0) { >> > - st->skip_samples = msc->start_pad = >> edit_list_media_time - curr_cts; >> > - >> > - // Shift the index entry timestamp by skip_samples >> to be correct. >> > - edit_list_dts_counter -= st->skip_samples; >> > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && >> st->codecpar->codec_id != AV_CODEC_ID_VORBIS && >> > + curr_cts < edit_list_media_time && curr_cts + >> frame_duration > edit_list_media_time && >> > + first_non_zero_audio_edit > 0) { >> > + packet_skip_samples = edit_list_media_time - >> curr_cts; >> > + st->skip_samples += packet_skip_samples; >> > + >> > + // Shift the index entry timestamp by >> packet_skip_samples to be correct. >> > + edit_list_dts_counter -= packet_skip_samples; >> > if (edit_list_start_encountered == 0) { >> > - edit_list_start_encountered = 1; >> > + edit_list_start_encountered = 1; >> > + // Make timestamps strictly monotonically >> increasing for audio, by rewriting timestamps for >> > + // discarded packets. >> > + if (frame_duration_buffer) { >> > + fix_index_entry_timestamps(st, >> st->nb_index_entries, edit_list_dts_counter, >> > + >> frame_duration_buffer, num_discarded_begin); >> > + av_freep(&frame_duration_buffer); >> > + } >> > } >> > >> > - av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio >> samples from curr_cts: %"PRId64"\n", st->skip_samples, curr_cts); >> > + av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio >> samples from curr_cts: %"PRId64"\n", packet_skip_samples, curr_cts); >> > } else { >> > flags |= AVINDEX_DISCARD_FRAME; >> > av_log(mov->fc, AV_LOG_DEBUG, "drop a frame at >> curr_cts: %"PRId64" @ %"PRId64"\n", curr_cts, index); >> > + >> > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO >> && edit_list_start_encountered == 0) { >> > + num_discarded_begin++; >> > + frame_duration_buffer = >> av_realloc(frame_duration_buffer, >> > + >> num_discarded_begin * sizeof(int64_t)); >> > + if (!frame_duration_buffer) { >> > + av_log(mov->fc, AV_LOG_ERROR, "Cannot >> reallocate frame duration buffer\n"); >> > + break; >> > + } >> > + frame_duration_buffer[num_discarded_begin - >> 1] = frame_duration; >> > + >> > + // Increment skip_samples for the first >> non-zero audio edit list >> > + if (first_non_zero_audio_edit > 0 && >> st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { >> > + st->skip_samples += frame_duration; >> > + msc->start_pad = st->skip_samples; >> > + } >> > + } >> > } >> > } else if (edit_list_start_encountered == 0) { >> > edit_list_start_encountered = 1; >> > + // Make timestamps strictly monotonically increasing >> for audio, by rewriting timestamps for >> > + // discarded packets. >> > + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && >> frame_duration_buffer) { >> > + fix_index_entry_timestamps(st, >> st->nb_index_entries, edit_list_dts_counter, >> > + frame_duration_buffer, >> num_discarded_begin); >> > + av_freep(&frame_duration_buffer); >> > + } >> > + >> > } >> > >> > if (add_index_entry(st, current->pos, >> edit_list_dts_counter, current->size, >> > diff --git a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac >> b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac >> > index 043c085..789681f 100644 >> > --- a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac >> > +++ b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac >> > @@ -7,7 +7,7 @@ duration_ts=103326 >> > start_time=0.000000 >> > duration=2.367000 >> > [/FORMAT] >> > -packet|pts=0|dts=0|duration=N/A >> > +packet|pts=-1024|dts=-1024|duration=1024 >> > packet|pts=0|dts=0|duration=1024 >> > packet|pts=1024|dts=1024|duration=1024 >> > packet|pts=2048|dts=2048|duration=1024 >> > diff --git a/tests/ref/fate/gaplessenc-pcm-to-mov-aac >> b/tests/ref/fate/gaplessenc-pcm-to-mov-aac >> > index 8b7e3f6..8702611 100644 >> > --- a/tests/ref/fate/gaplessenc-pcm-to-mov-aac >> > +++ b/tests/ref/fate/gaplessenc-pcm-to-mov-aac >> > @@ -7,7 +7,7 @@ duration_ts=529200 >> > start_time=0.000000 >> > duration=12.024000 >> > [/FORMAT] >> > -packet|pts=0|dts=0|duration=N/A >> > +packet|pts=-1024|dts=-1024|duration=1024 >> > packet|pts=0|dts=0|duration=1024 >> > packet|pts=1024|dts=1024|duration=1024 >> > packet|pts=2048|dts=2048|duration=1024 >> >> This is a complex patch, and builds upon new code that isn't quite >> known to me, the result looks like an improvement to me. >> >> Does it work with the "skip_manual" libavcodec option? >> >> Nice catch. sent the patch again to make it work with skip_manual > > I also think the fate tests should be updated to include the packet >> flags, since they are just as important as the timestamps. >> > > I had another patch making ffprobe show the DISCARD flag. If / once that > patch is applied it will be easier to modify the fate test for gapless to > show the discard flag. > > >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> > >
diff --git a/libavcodec/utils.c b/libavcodec/utils.c index b0345b6..e18476c 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -2320,7 +2320,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, uint32_t discard_padding = 0; uint8_t skip_reason = 0; uint8_t discard_reason = 0; - int demuxer_skip_samples = 0; // copy to ensure we do not change avpkt AVPacket tmp = *avpkt; int did_split = av_packet_split_side_data(&tmp); @@ -2328,7 +2327,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, if (ret < 0) goto fail; - demuxer_skip_samples = avctx->internal->skip_samples; avctx->internal->pkt = &tmp; if (HAVE_THREADS && avctx->active_thread_type & FF_THREAD_FRAME) ret = ff_thread_decode_frame(avctx, frame, got_frame_ptr, &tmp); @@ -2353,13 +2351,6 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, frame->sample_rate = avctx->sample_rate; } - - if (frame->flags & AV_FRAME_FLAG_DISCARD) { - // If using discard frame flag, ignore skip_samples set by the decoder. - avctx->internal->skip_samples = demuxer_skip_samples; - *got_frame_ptr = 0; - } - side= av_packet_get_side_data(avctx->internal->pkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size); if(side && side_size>=10) { avctx->internal->skip_samples = AV_RL32(side); @@ -2369,6 +2360,12 @@ int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx, skip_reason = AV_RL8(side + 8); discard_reason = AV_RL8(side + 9); } + + if ((frame->flags & AV_FRAME_FLAG_DISCARD) && *got_frame_ptr) { + avctx->internal->skip_samples -= frame->nb_samples; + *got_frame_ptr = 0; + } + if (avctx->internal->skip_samples > 0 && *got_frame_ptr && !(avctx->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) { if(frame->nb_samples <= avctx->internal->skip_samples){ diff --git a/libavformat/mov.c b/libavformat/mov.c index b84d9c0..bb86780 100644 --- a/libavformat/mov.c +++ b/libavformat/mov.c @@ -2856,6 +2856,21 @@ static int64_t add_index_entry(AVStream *st, int64_t pos, int64_t timestamp, } /** + * Rewrite timestamps of index entries in the range [end_index - frame_duration_buffer_size, end_index) + * by subtracting end_ts successively by the amounts given in frame_duration_buffer. + */ +static void fix_index_entry_timestamps(AVStream* st, int end_index, int64_t end_ts, + int64_t* frame_duration_buffer, + int frame_duration_buffer_size) { + int i = 0; + av_assert0(end_index >= 0 && end_index <= st->nb_index_entries); + for (i = 0; i < frame_duration_buffer_size; i++) { + end_ts -= frame_duration_buffer[frame_duration_buffer_size - 1 - i]; + st->index_entries[end_index - 1 - i].timestamp = end_ts; + } +} + +/** * Append a new ctts entry to ctts_data. * Returns the new ctts_count if successful, else returns -1. */ @@ -2919,7 +2934,10 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) int64_t edit_list_media_time_dts = 0; int64_t edit_list_start_encountered = 0; int64_t search_timestamp = 0; - + int64_t* frame_duration_buffer = NULL; + int num_discarded_begin = 0; + int first_non_zero_audio_edit = -1; + int packet_skip_samples = 0; if (!msc->elst_data || msc->elst_count <= 0) { return; @@ -2955,6 +2973,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) edit_list_index++; edit_list_dts_counter = edit_list_dts_entry_end; edit_list_dts_entry_end += edit_list_duration; + num_discarded_begin = 0; if (edit_list_media_time == -1) { continue; } @@ -2962,7 +2981,14 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) // If we encounter a non-negative edit list reset the skip_samples/start_pad fields and set them // according to the edit list below. if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - st->skip_samples = msc->start_pad = 0; + if (first_non_zero_audio_edit < 0) { + first_non_zero_audio_edit = 1; + } else { + first_non_zero_audio_edit = 0; + } + + if (first_non_zero_audio_edit > 0) + st->skip_samples = msc->start_pad = 0; } //find closest previous key frame @@ -3041,24 +3067,57 @@ static void mov_fix_index(MOVContext *mov, AVStream *st) } if (curr_cts < edit_list_media_time || curr_cts >= (edit_list_duration + edit_list_media_time)) { - if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && curr_cts < edit_list_media_time && - curr_cts + frame_duration > edit_list_media_time && - st->skip_samples == 0 && msc->start_pad == 0) { - st->skip_samples = msc->start_pad = edit_list_media_time - curr_cts; - - // Shift the index entry timestamp by skip_samples to be correct. - edit_list_dts_counter -= st->skip_samples; + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && st->codecpar->codec_id != AV_CODEC_ID_VORBIS && + curr_cts < edit_list_media_time && curr_cts + frame_duration > edit_list_media_time && + first_non_zero_audio_edit > 0) { + packet_skip_samples = edit_list_media_time - curr_cts; + st->skip_samples += packet_skip_samples; + + // Shift the index entry timestamp by packet_skip_samples to be correct. + edit_list_dts_counter -= packet_skip_samples; if (edit_list_start_encountered == 0) { - edit_list_start_encountered = 1; + edit_list_start_encountered = 1; + // Make timestamps strictly monotonically increasing for audio, by rewriting timestamps for + // discarded packets. + if (frame_duration_buffer) { + fix_index_entry_timestamps(st, st->nb_index_entries, edit_list_dts_counter, + frame_duration_buffer, num_discarded_begin); + av_freep(&frame_duration_buffer); + } } - av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples from curr_cts: %"PRId64"\n", st->skip_samples, curr_cts); + av_log(mov->fc, AV_LOG_DEBUG, "skip %d audio samples from curr_cts: %"PRId64"\n", packet_skip_samples, curr_cts); } else { flags |= AVINDEX_DISCARD_FRAME; av_log(mov->fc, AV_LOG_DEBUG, "drop a frame at curr_cts: %"PRId64" @ %"PRId64"\n", curr_cts, index); + + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && edit_list_start_encountered == 0) { + num_discarded_begin++; + frame_duration_buffer = av_realloc(frame_duration_buffer, + num_discarded_begin * sizeof(int64_t)); + if (!frame_duration_buffer) { + av_log(mov->fc, AV_LOG_ERROR, "Cannot reallocate frame duration buffer\n"); + break; + } + frame_duration_buffer[num_discarded_begin - 1] = frame_duration; + + // Increment skip_samples for the first non-zero audio edit list + if (first_non_zero_audio_edit > 0 && st->codecpar->codec_id != AV_CODEC_ID_VORBIS) { + st->skip_samples += frame_duration; + msc->start_pad = st->skip_samples; + } + } } } else if (edit_list_start_encountered == 0) { edit_list_start_encountered = 1; + // Make timestamps strictly monotonically increasing for audio, by rewriting timestamps for + // discarded packets. + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO && frame_duration_buffer) { + fix_index_entry_timestamps(st, st->nb_index_entries, edit_list_dts_counter, + frame_duration_buffer, num_discarded_begin); + av_freep(&frame_duration_buffer); + } + } if (add_index_entry(st, current->pos, edit_list_dts_counter, current->size, diff --git a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac index 043c085..789681f 100644 --- a/tests/ref/fate/gaplessenc-itunes-to-ipod-aac +++ b/tests/ref/fate/gaplessenc-itunes-to-ipod-aac @@ -7,7 +7,7 @@ duration_ts=103326 start_time=0.000000 duration=2.367000 [/FORMAT] -packet|pts=0|dts=0|duration=N/A +packet|pts=-1024|dts=-1024|duration=1024 packet|pts=0|dts=0|duration=1024 packet|pts=1024|dts=1024|duration=1024 packet|pts=2048|dts=2048|duration=1024 diff --git a/tests/ref/fate/gaplessenc-pcm-to-mov-aac b/tests/ref/fate/gaplessenc-pcm-to-mov-aac index 8b7e3f6..8702611 100644 --- a/tests/ref/fate/gaplessenc-pcm-to-mov-aac +++ b/tests/ref/fate/gaplessenc-pcm-to-mov-aac @@ -7,7 +7,7 @@ duration_ts=529200 start_time=0.000000 duration=12.024000 [/FORMAT] -packet|pts=0|dts=0|duration=N/A +packet|pts=-1024|dts=-1024|duration=1024 packet|pts=0|dts=0|duration=1024 packet|pts=1024|dts=1024|duration=1024 packet|pts=2048|dts=2048|duration=1024
Signed-off-by: Sasi Inguva <isasi@google.com> --- libavcodec/utils.c | 15 +++--- libavformat/mov.c | 81 ++++++++++++++++++++++++---- tests/ref/fate/gaplessenc-itunes-to-ipod-aac | 2 +- tests/ref/fate/gaplessenc-pcm-to-mov-aac | 2 +- 4 files changed, 78 insertions(+), 22 deletions(-)