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?
I also think the fate tests should be updated to include the packet
flags, since they are just as important as the timestamps.
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