On Thu, Sep 22, 2016 at 5:49 AM, wm4 <nfx...@googlemail.com> wrote:
> On Tue, 20 Sep 2016 14:29:46 -0700
> Sasi Inguva <isasi-at-google....@ffmpeg.org> wrote:
>
> > Signed-off-by: Sasi Inguva <is...@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.
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>
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