On 10/17/22, Hendrik Leppkes <h.lepp...@gmail.com> wrote:
> On Thu, Sep 8, 2022 at 10:26 AM <jyrk...@nekonyansoft.com> wrote:
>>
>> From: Jyrki Vesterinen <jyrk...@nekonyansoft.com>
>>
>> If a developer using FFmpeg libraries seeks into an earlier position and
>> calls
>> avcodec_flush_buffers() afterwards as recommended, the Vorbis decoder will
>> drop
>> the next frame, since buffer flushing clears the first_frame flag. As a
>> result,
>> the audio samples the calling code receives may be ahead of the requested
>> seek
>> position, which is unacceptable in some use cases such as playing a
>> looping
>> sound effect.
>>
>> This commit removes the first_frame flag entirely and instead uses the
>> presentation timestamp to determine if it's the first frame.
>> ---
>> libavcodec/vorbisdec.c | 5 +----
>> 1 file changed, 1 insertion(+), 4 deletions(-)
>>
>> diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
>> index 4d03947c49..d4b030d7b9 100644
>> --- a/libavcodec/vorbisdec.c
>> +++ b/libavcodec/vorbisdec.c
>> @@ -130,7 +130,6 @@ typedef struct vorbis_context_s {
>> AVFloatDSPContext *fdsp;
>>
>> FFTContext mdct[2];
>> - uint8_t first_frame;
>> uint32_t version;
>> uint8_t audio_channels;
>> uint32_t audio_samplerate;
>> @@ -1845,8 +1844,7 @@ static int vorbis_decode_frame(AVCodecContext
>> *avctx, AVFrame *frame,
>> if ((len = vorbis_parse_audio_packet(vc, channel_ptrs)) <= 0)
>> return len;
>>
>> - if (!vc->first_frame) {
>> - vc->first_frame = 1;
>> + if (frame->pts < 0) {
>> *got_frame_ptr = 0;
>> av_frame_unref(frame);
>> return buf_size;
>> @@ -1881,7 +1879,6 @@ static av_cold void
>> vorbis_decode_flush(AVCodecContext *avctx)
>> sizeof(*vc->saved));
>> }
>> vc->previous_window = -1;
>> - vc->first_frame = 0;
>> }
>>
>> const FFCodec ff_vorbis_decoder = {
>> --
>> 2.37.2.windows.2
>>
>
> This change seems to be rather fragile and faulty, causing vorbis
> decoding to fail in various scenarios for a bunch of downstream
> projects.
>
> - A user may not set pts at all, resulting in all frames being dropped
> (pure audio files don't necessarily need timestamps)
> - A seek could happen before any frame is ever decoded, resulting in
> wrong drops, potentially in the middle of playback if the user seeks
> backwards after opening in the middle.
>
> In general, using timestamps to control decoder behavior is often just
> wrong, as timestamps are not reliable, and most importantly, not tied
> to the bitstream at all.
>
> Can we revert this and re-think the approach?
Are you saying that previous solution was better than current one?
By your own words its ever worse that current state.
>
> - Hendrik
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