* [FFmpeg-devel] [PATCH 1/2] libfdk-aacdec: Add an option for setting the decoder's DRC album mode @ 2022-01-19 20:38 Martin Storsjö 2022-01-19 20:38 ` [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end Martin Storsjö 0 siblings, 1 reply; 6+ messages in thread From: Martin Storsjö @ 2022-01-19 20:38 UTC (permalink / raw) To: ffmpeg-devel --- libavcodec/libfdk-aacdec.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c index ffa1fdcce3..93b52023b0 100644 --- a/libavcodec/libfdk-aacdec.c +++ b/libavcodec/libfdk-aacdec.c @@ -56,6 +56,7 @@ typedef struct FDKAACDecContext { int drc_heavy; int drc_effect; int drc_cut; + int album_mode; int level_limit; int output_delay; } FDKAACDecContext; @@ -87,6 +88,10 @@ static const AVOption fdk_aac_dec_options[] = { #if FDKDEC_VER_AT_LEAST(3, 0) // 3.0.0 { "drc_effect","Dynamic Range Control: effect type, where e.g. [0] is none and [6] is general", OFFSET(drc_effect), AV_OPT_TYPE_INT, { .i64 = -1}, -1, 8, AD, NULL }, +#endif +#if FDKDEC_VER_AT_LEAST(3, 1) // 3.1.0 + { "album_mode","Dynamic Range Control: album mode, where [0] is off and [1] is on", + OFFSET(album_mode), AV_OPT_TYPE_INT, { .i64 = -1}, -1, 1, AD, NULL }, #endif { NULL } }; @@ -335,6 +340,15 @@ static av_cold int fdk_aac_decode_init(AVCodecContext *avctx) } #endif +#if FDKDEC_VER_AT_LEAST(3, 1) // 3.1.0 + if (s->album_mode != -1) { + if (aacDecoder_SetParam(s->handle, AAC_UNIDRC_ALBUM_MODE, s->album_mode) != AAC_DEC_OK) { + av_log(avctx, AV_LOG_ERROR, "Unable to set album mode in the decoder\n"); + return AVERROR_UNKNOWN; + } + } +#endif + avctx->sample_fmt = AV_SAMPLE_FMT_S16; s->decoder_buffer_size = DECODER_BUFFSIZE * DECODER_MAX_CHANNELS; -- 2.32.0 (Apple Git-132) _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
* [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end 2022-01-19 20:38 [FFmpeg-devel] [PATCH 1/2] libfdk-aacdec: Add an option for setting the decoder's DRC album mode Martin Storsjö @ 2022-01-19 20:38 ` Martin Storsjö 2022-01-21 13:00 ` Andreas Rheinhardt 0 siblings, 1 reply; 6+ messages in thread From: Martin Storsjö @ 2022-01-19 20:38 UTC (permalink / raw) To: ffmpeg-devel Also trim off delay samples at the start instead of adjusting pts to compensate for them; this avoids unwanted offsets if working with raw samples without considering their pts. --- libavcodec/libfdk-aacdec.c | 80 +++++++++++++++++++++++++++++++------- 1 file changed, 65 insertions(+), 15 deletions(-) diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c index 93b52023b0..d560e313ca 100644 --- a/libavcodec/libfdk-aacdec.c +++ b/libavcodec/libfdk-aacdec.c @@ -58,7 +58,11 @@ typedef struct FDKAACDecContext { int drc_cut; int album_mode; int level_limit; - int output_delay; +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 + int output_delay_set; + int flush_samples; + int delay_samples; +#endif } FDKAACDecContext; @@ -123,7 +127,12 @@ static int get_stream_info(AVCodecContext *avctx) avctx->sample_rate = info->sampleRate; avctx->frame_size = info->frameSize; #if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 - s->output_delay = info->outputDelay; + if (!s->output_delay_set && info->outputDelay) { + // Set this only once. + s->flush_samples = info->outputDelay; + s->delay_samples = info->outputDelay; + s->output_delay_set = 1; + } #endif for (i = 0; i < info->numChannels; i++) { @@ -367,14 +376,31 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, int ret; AAC_DECODER_ERROR err; UINT valid = avpkt->size; + UINT flags = 0; + int input_offset = 0; - err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); - if (err != AAC_DEC_OK) { - av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); - return AVERROR_INVALIDDATA; + if (avpkt->size) { + err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); + if (err != AAC_DEC_OK) { + av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); + return AVERROR_INVALIDDATA; + } + } else { +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 + /* Handle decoder draining */ + if (s->flush_samples > 0) { + flags |= AACDEC_FLUSH; + } else { + return AVERROR_EOF; + } +#else + return AVERROR_EOF; +#endif } - err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, s->decoder_buffer_size / sizeof(INT_PCM), 0); + err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, + s->decoder_buffer_size / sizeof(INT_PCM), + flags); if (err == AAC_DEC_NOT_ENOUGH_BITS) { ret = avpkt->size - valid; goto end; @@ -390,16 +416,36 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, goto end; frame->nb_samples = avctx->frame_size; +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 + if (flags & AACDEC_FLUSH) { + // Only return the right amount of samples at the end; if calling the + // decoder with AACDEC_FLUSH, it will keep returning frames indefinitely. + frame->nb_samples = FFMIN(s->flush_samples, frame->nb_samples); + av_log(s, AV_LOG_DEBUG, "Returning %d/%d delayed samples.\n", + frame->nb_samples, s->flush_samples); + s->flush_samples -= frame->nb_samples; + } else { + // Trim off samples from the start to compensate for extra decoder + // delay. We could also just adjust the pts, but this avoids + // including the extra samples in the output altogether. + if (s->delay_samples) { + int drop_samples = FFMIN(s->delay_samples, frame->nb_samples); + av_log(s, AV_LOG_DEBUG, "Dropping %d/%d delayed samples.\n", + drop_samples, s->delay_samples); + s->delay_samples -= drop_samples; + frame->nb_samples -= drop_samples; + input_offset = drop_samples * avctx->channels; + if (frame->nb_samples <= 0) + return 0; + } + } +#endif + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) goto end; - if (frame->pts != AV_NOPTS_VALUE) - frame->pts -= av_rescale_q(s->output_delay, - (AVRational){1, avctx->sample_rate}, - avctx->time_base); - - memcpy(frame->extended_data[0], s->decoder_buffer, - avctx->channels * avctx->frame_size * + memcpy(frame->extended_data[0], s->decoder_buffer + input_offset, + avctx->channels * frame->nb_samples * av_get_bytes_per_sample(avctx->sample_fmt)); *got_frame_ptr = 1; @@ -432,7 +478,11 @@ const AVCodec ff_libfdk_aac_decoder = { .decode = fdk_aac_decode_frame, .close = fdk_aac_decode_close, .flush = fdk_aac_decode_flush, - .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 + | AV_CODEC_CAP_DELAY +#endif + , .priv_class = &fdk_aac_dec_class, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, -- 2.32.0 (Apple Git-132) _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end 2022-01-19 20:38 ` [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end Martin Storsjö @ 2022-01-21 13:00 ` Andreas Rheinhardt 2022-01-21 13:58 ` Martin Storsjö 0 siblings, 1 reply; 6+ messages in thread From: Andreas Rheinhardt @ 2022-01-21 13:00 UTC (permalink / raw) To: ffmpeg-devel Martin Storsjö: > Also trim off delay samples at the start instead of adjusting pts > to compensate for them; this avoids unwanted offsets if working > with raw samples without considering their pts. > --- > libavcodec/libfdk-aacdec.c | 80 +++++++++++++++++++++++++++++++------- > 1 file changed, 65 insertions(+), 15 deletions(-) > > diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c > index 93b52023b0..d560e313ca 100644 > --- a/libavcodec/libfdk-aacdec.c > +++ b/libavcodec/libfdk-aacdec.c > @@ -58,7 +58,11 @@ typedef struct FDKAACDecContext { > int drc_cut; > int album_mode; > int level_limit; > - int output_delay; > +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 > + int output_delay_set; > + int flush_samples; > + int delay_samples; > +#endif > } FDKAACDecContext; > > > @@ -123,7 +127,12 @@ static int get_stream_info(AVCodecContext *avctx) > avctx->sample_rate = info->sampleRate; > avctx->frame_size = info->frameSize; > #if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 > - s->output_delay = info->outputDelay; > + if (!s->output_delay_set && info->outputDelay) { > + // Set this only once. > + s->flush_samples = info->outputDelay; > + s->delay_samples = info->outputDelay; > + s->output_delay_set = 1; > + } > #endif > > for (i = 0; i < info->numChannels; i++) { > @@ -367,14 +376,31 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, > int ret; > AAC_DECODER_ERROR err; > UINT valid = avpkt->size; > + UINT flags = 0; > + int input_offset = 0; > > - err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); > - if (err != AAC_DEC_OK) { > - av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); > - return AVERROR_INVALIDDATA; > + if (avpkt->size) { > + err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); > + if (err != AAC_DEC_OK) { > + av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); > + return AVERROR_INVALIDDATA; > + } > + } else { > +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 > + /* Handle decoder draining */ > + if (s->flush_samples > 0) { > + flags |= AACDEC_FLUSH; > + } else { > + return AVERROR_EOF; > + } > +#else > + return AVERROR_EOF; > +#endif > } > > - err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, s->decoder_buffer_size / sizeof(INT_PCM), 0); > + err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, > + s->decoder_buffer_size / sizeof(INT_PCM), > + flags); > if (err == AAC_DEC_NOT_ENOUGH_BITS) { > ret = avpkt->size - valid; > goto end; > @@ -390,16 +416,36 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, > goto end; > frame->nb_samples = avctx->frame_size; > > +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 > + if (flags & AACDEC_FLUSH) { > + // Only return the right amount of samples at the end; if calling the > + // decoder with AACDEC_FLUSH, it will keep returning frames indefinitely. > + frame->nb_samples = FFMIN(s->flush_samples, frame->nb_samples); > + av_log(s, AV_LOG_DEBUG, "Returning %d/%d delayed samples.\n", > + frame->nb_samples, s->flush_samples); > + s->flush_samples -= frame->nb_samples; > + } else { > + // Trim off samples from the start to compensate for extra decoder > + // delay. We could also just adjust the pts, but this avoids > + // including the extra samples in the output altogether. > + if (s->delay_samples) { > + int drop_samples = FFMIN(s->delay_samples, frame->nb_samples); > + av_log(s, AV_LOG_DEBUG, "Dropping %d/%d delayed samples.\n", > + drop_samples, s->delay_samples); > + s->delay_samples -= drop_samples; > + frame->nb_samples -= drop_samples; > + input_offset = drop_samples * avctx->channels; > + if (frame->nb_samples <= 0) > + return 0; > + } > + } > +#endif > + > if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) > goto end; > > - if (frame->pts != AV_NOPTS_VALUE) > - frame->pts -= av_rescale_q(s->output_delay, > - (AVRational){1, avctx->sample_rate}, > - avctx->time_base); > - > - memcpy(frame->extended_data[0], s->decoder_buffer, > - avctx->channels * avctx->frame_size * > + memcpy(frame->extended_data[0], s->decoder_buffer + input_offset, > + avctx->channels * frame->nb_samples * > av_get_bytes_per_sample(avctx->sample_fmt)); > > *got_frame_ptr = 1; > @@ -432,7 +478,11 @@ const AVCodec ff_libfdk_aac_decoder = { > .decode = fdk_aac_decode_frame, > .close = fdk_aac_decode_close, > .flush = fdk_aac_decode_flush, > - .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, > + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF > +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 > + | AV_CODEC_CAP_DELAY > +#endif > + , > .priv_class = &fdk_aac_dec_class, > .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | > FF_CODEC_CAP_INIT_CLEANUP, > When I use the libfdk-aac decoder I get the exact number of samples like with the native aac decoder (namely number of frames * 1024, as expected). What makes you believe this is necessary? - Andreas _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end 2022-01-21 13:00 ` Andreas Rheinhardt @ 2022-01-21 13:58 ` Martin Storsjö 2022-01-21 14:42 ` Andreas Rheinhardt 0 siblings, 1 reply; 6+ messages in thread From: Martin Storsjö @ 2022-01-21 13:58 UTC (permalink / raw) To: FFmpeg development discussions and patches On Fri, 21 Jan 2022, Andreas Rheinhardt wrote: > Martin Storsjö: >> Also trim off delay samples at the start instead of adjusting pts >> to compensate for them; this avoids unwanted offsets if working >> with raw samples without considering their pts. >> --- >> libavcodec/libfdk-aacdec.c | 80 +++++++++++++++++++++++++++++++------- >> 1 file changed, 65 insertions(+), 15 deletions(-) >> >> diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c >> index 93b52023b0..d560e313ca 100644 >> --- a/libavcodec/libfdk-aacdec.c >> +++ b/libavcodec/libfdk-aacdec.c >> @@ -58,7 +58,11 @@ typedef struct FDKAACDecContext { >> int drc_cut; >> int album_mode; >> int level_limit; >> - int output_delay; >> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >> + int output_delay_set; >> + int flush_samples; >> + int delay_samples; >> +#endif >> } FDKAACDecContext; >> >> >> @@ -123,7 +127,12 @@ static int get_stream_info(AVCodecContext *avctx) >> avctx->sample_rate = info->sampleRate; >> avctx->frame_size = info->frameSize; >> #if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >> - s->output_delay = info->outputDelay; >> + if (!s->output_delay_set && info->outputDelay) { >> + // Set this only once. >> + s->flush_samples = info->outputDelay; >> + s->delay_samples = info->outputDelay; >> + s->output_delay_set = 1; >> + } >> #endif >> >> for (i = 0; i < info->numChannels; i++) { >> @@ -367,14 +376,31 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, >> int ret; >> AAC_DECODER_ERROR err; >> UINT valid = avpkt->size; >> + UINT flags = 0; >> + int input_offset = 0; >> >> - err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); >> - if (err != AAC_DEC_OK) { >> - av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); >> - return AVERROR_INVALIDDATA; >> + if (avpkt->size) { >> + err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, &valid); >> + if (err != AAC_DEC_OK) { >> + av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: %x\n", err); >> + return AVERROR_INVALIDDATA; >> + } >> + } else { >> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >> + /* Handle decoder draining */ >> + if (s->flush_samples > 0) { >> + flags |= AACDEC_FLUSH; >> + } else { >> + return AVERROR_EOF; >> + } >> +#else >> + return AVERROR_EOF; >> +#endif >> } >> >> - err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, s->decoder_buffer_size / sizeof(INT_PCM), 0); >> + err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) s->decoder_buffer, >> + s->decoder_buffer_size / sizeof(INT_PCM), >> + flags); >> if (err == AAC_DEC_NOT_ENOUGH_BITS) { >> ret = avpkt->size - valid; >> goto end; >> @@ -390,16 +416,36 @@ static int fdk_aac_decode_frame(AVCodecContext *avctx, void *data, >> goto end; >> frame->nb_samples = avctx->frame_size; >> >> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >> + if (flags & AACDEC_FLUSH) { >> + // Only return the right amount of samples at the end; if calling the >> + // decoder with AACDEC_FLUSH, it will keep returning frames indefinitely. >> + frame->nb_samples = FFMIN(s->flush_samples, frame->nb_samples); >> + av_log(s, AV_LOG_DEBUG, "Returning %d/%d delayed samples.\n", >> + frame->nb_samples, s->flush_samples); >> + s->flush_samples -= frame->nb_samples; >> + } else { >> + // Trim off samples from the start to compensate for extra decoder >> + // delay. We could also just adjust the pts, but this avoids >> + // including the extra samples in the output altogether. >> + if (s->delay_samples) { >> + int drop_samples = FFMIN(s->delay_samples, frame->nb_samples); >> + av_log(s, AV_LOG_DEBUG, "Dropping %d/%d delayed samples.\n", >> + drop_samples, s->delay_samples); >> + s->delay_samples -= drop_samples; >> + frame->nb_samples -= drop_samples; >> + input_offset = drop_samples * avctx->channels; >> + if (frame->nb_samples <= 0) >> + return 0; >> + } >> + } >> +#endif >> + >> if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) >> goto end; >> >> - if (frame->pts != AV_NOPTS_VALUE) >> - frame->pts -= av_rescale_q(s->output_delay, >> - (AVRational){1, avctx->sample_rate}, >> - avctx->time_base); >> - >> - memcpy(frame->extended_data[0], s->decoder_buffer, >> - avctx->channels * avctx->frame_size * >> + memcpy(frame->extended_data[0], s->decoder_buffer + input_offset, >> + avctx->channels * frame->nb_samples * >> av_get_bytes_per_sample(avctx->sample_fmt)); >> >> *got_frame_ptr = 1; >> @@ -432,7 +478,11 @@ const AVCodec ff_libfdk_aac_decoder = { >> .decode = fdk_aac_decode_frame, >> .close = fdk_aac_decode_close, >> .flush = fdk_aac_decode_flush, >> - .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, >> + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF >> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >> + | AV_CODEC_CAP_DELAY >> +#endif >> + , >> .priv_class = &fdk_aac_dec_class, >> .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | >> FF_CODEC_CAP_INIT_CLEANUP, >> > > When I use the libfdk-aac decoder I get the exact number of samples like > with the native aac decoder (namely number of frames * 1024, as > expected). What makes you believe this is necessary? The fdk-aac decoder can have, depending on combination of options, some amount of extra internal delay, that the libavcodec internal aac decoder doesn't have. (It's also possible to set the options in a state where the fdk-aac decoder doesn't induce any extra delay.) Currently, we compensate for that extra delay by just offsetting pts backwards, so for a stream with N packets, we return samples with timestamps [-delay,N*framesize-delay]. In order not to lose data at the end, we must make the decoder flushable and flush up to (delay) samples at the end. And since one doesn't normally expect extra delay samples at the start of an AAC decoder output, we also trim out the same amount of samples at the start (to simplify for users that don't observe the pts, who otherwise are surprised by the stream starting from pts -delay instead of at pts 0). // Martin _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end 2022-01-21 13:58 ` Martin Storsjö @ 2022-01-21 14:42 ` Andreas Rheinhardt 2022-02-03 11:53 ` Martin Storsjö 0 siblings, 1 reply; 6+ messages in thread From: Andreas Rheinhardt @ 2022-01-21 14:42 UTC (permalink / raw) To: ffmpeg-devel Martin Storsjö: > On Fri, 21 Jan 2022, Andreas Rheinhardt wrote: > >> Martin Storsjö: >>> Also trim off delay samples at the start instead of adjusting pts >>> to compensate for them; this avoids unwanted offsets if working >>> with raw samples without considering their pts. >>> --- >>> libavcodec/libfdk-aacdec.c | 80 +++++++++++++++++++++++++++++++------- >>> 1 file changed, 65 insertions(+), 15 deletions(-) >>> >>> diff --git a/libavcodec/libfdk-aacdec.c b/libavcodec/libfdk-aacdec.c >>> index 93b52023b0..d560e313ca 100644 >>> --- a/libavcodec/libfdk-aacdec.c >>> +++ b/libavcodec/libfdk-aacdec.c >>> @@ -58,7 +58,11 @@ typedef struct FDKAACDecContext { >>> int drc_cut; >>> int album_mode; >>> int level_limit; >>> - int output_delay; >>> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >>> + int output_delay_set; >>> + int flush_samples; >>> + int delay_samples; >>> +#endif >>> } FDKAACDecContext; >>> >>> >>> @@ -123,7 +127,12 @@ static int get_stream_info(AVCodecContext *avctx) >>> avctx->sample_rate = info->sampleRate; >>> avctx->frame_size = info->frameSize; >>> #if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >>> - s->output_delay = info->outputDelay; >>> + if (!s->output_delay_set && info->outputDelay) { >>> + // Set this only once. >>> + s->flush_samples = info->outputDelay; >>> + s->delay_samples = info->outputDelay; >>> + s->output_delay_set = 1; >>> + } >>> #endif >>> >>> for (i = 0; i < info->numChannels; i++) { >>> @@ -367,14 +376,31 @@ static int fdk_aac_decode_frame(AVCodecContext >>> *avctx, void *data, >>> int ret; >>> AAC_DECODER_ERROR err; >>> UINT valid = avpkt->size; >>> + UINT flags = 0; >>> + int input_offset = 0; >>> >>> - err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, >>> &valid); >>> - if (err != AAC_DEC_OK) { >>> - av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: >>> %x\n", err); >>> - return AVERROR_INVALIDDATA; >>> + if (avpkt->size) { >>> + err = aacDecoder_Fill(s->handle, &avpkt->data, &avpkt->size, >>> &valid); >>> + if (err != AAC_DEC_OK) { >>> + av_log(avctx, AV_LOG_ERROR, "aacDecoder_Fill() failed: >>> %x\n", err); >>> + return AVERROR_INVALIDDATA; >>> + } >>> + } else { >>> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >>> + /* Handle decoder draining */ >>> + if (s->flush_samples > 0) { >>> + flags |= AACDEC_FLUSH; >>> + } else { >>> + return AVERROR_EOF; >>> + } >>> +#else >>> + return AVERROR_EOF; >>> +#endif >>> } >>> >>> - err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) >>> s->decoder_buffer, s->decoder_buffer_size / sizeof(INT_PCM), 0); >>> + err = aacDecoder_DecodeFrame(s->handle, (INT_PCM *) >>> s->decoder_buffer, >>> + s->decoder_buffer_size / >>> sizeof(INT_PCM), >>> + flags); >>> if (err == AAC_DEC_NOT_ENOUGH_BITS) { >>> ret = avpkt->size - valid; >>> goto end; >>> @@ -390,16 +416,36 @@ static int fdk_aac_decode_frame(AVCodecContext >>> *avctx, void *data, >>> goto end; >>> frame->nb_samples = avctx->frame_size; >>> >>> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >>> + if (flags & AACDEC_FLUSH) { >>> + // Only return the right amount of samples at the end; if >>> calling the >>> + // decoder with AACDEC_FLUSH, it will keep returning frames >>> indefinitely. >>> + frame->nb_samples = FFMIN(s->flush_samples, frame->nb_samples); >>> + av_log(s, AV_LOG_DEBUG, "Returning %d/%d delayed samples.\n", >>> + frame->nb_samples, s->flush_samples); >>> + s->flush_samples -= frame->nb_samples; >>> + } else { >>> + // Trim off samples from the start to compensate for extra >>> decoder >>> + // delay. We could also just adjust the pts, but this avoids >>> + // including the extra samples in the output altogether. >>> + if (s->delay_samples) { >>> + int drop_samples = FFMIN(s->delay_samples, >>> frame->nb_samples); >>> + av_log(s, AV_LOG_DEBUG, "Dropping %d/%d delayed >>> samples.\n", >>> + drop_samples, s->delay_samples); >>> + s->delay_samples -= drop_samples; >>> + frame->nb_samples -= drop_samples; >>> + input_offset = drop_samples * avctx->channels; >>> + if (frame->nb_samples <= 0) >>> + return 0; >>> + } >>> + } >>> +#endif >>> + >>> if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) >>> goto end; >>> >>> - if (frame->pts != AV_NOPTS_VALUE) >>> - frame->pts -= av_rescale_q(s->output_delay, >>> - (AVRational){1, avctx->sample_rate}, >>> - avctx->time_base); >>> - >>> - memcpy(frame->extended_data[0], s->decoder_buffer, >>> - avctx->channels * avctx->frame_size * >>> + memcpy(frame->extended_data[0], s->decoder_buffer + input_offset, >>> + avctx->channels * frame->nb_samples * >>> av_get_bytes_per_sample(avctx->sample_fmt)); >>> >>> *got_frame_ptr = 1; >>> @@ -432,7 +478,11 @@ const AVCodec ff_libfdk_aac_decoder = { >>> .decode = fdk_aac_decode_frame, >>> .close = fdk_aac_decode_close, >>> .flush = fdk_aac_decode_flush, >>> - .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, >>> + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF >>> +#if FDKDEC_VER_AT_LEAST(2, 5) // 2.5.10 >>> + | AV_CODEC_CAP_DELAY >>> +#endif >>> + , >>> .priv_class = &fdk_aac_dec_class, >>> .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | >>> FF_CODEC_CAP_INIT_CLEANUP, >>> >> >> When I use the libfdk-aac decoder I get the exact number of samples like >> with the native aac decoder (namely number of frames * 1024, as >> expected). What makes you believe this is necessary? > > The fdk-aac decoder can have, depending on combination of options, some > amount of extra internal delay, that the libavcodec internal aac decoder > doesn't have. (It's also possible to set the options in a state where > the fdk-aac decoder doesn't induce any extra delay.) > > Currently, we compensate for that extra delay by just offsetting pts > backwards, so for a stream with N packets, we return samples with > timestamps [-delay,N*framesize-delay]. > > In order not to lose data at the end, we must make the decoder flushable > and flush up to (delay) samples at the end. And since one doesn't > normally expect extra delay samples at the start of an AAC decoder > output, we also trim out the same amount of samples at the start (to > simplify for users that don't observe the pts, who otherwise are > surprised by the stream starting from pts -delay instead of at pts 0). > Interesting: There is indeed a delay at the start (720 samples in a quick test) compared to the native AAC decoder. Furthermore, the current code is buggy, as it believes that avcodec->time_base to be the time_base of the returned AVFrames (it is in reality avcodec->pkt_timebase; just test with AAC-in-Matroska for this). I haven't tested your patches, but I have now realized that there is indeed an issue. And your patch should also fix the wrong timebase issue. - Andreas _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end 2022-01-21 14:42 ` Andreas Rheinhardt @ 2022-02-03 11:53 ` Martin Storsjö 0 siblings, 0 replies; 6+ messages in thread From: Martin Storsjö @ 2022-02-03 11:53 UTC (permalink / raw) To: FFmpeg development discussions and patches On Fri, 21 Jan 2022, Andreas Rheinhardt wrote: > Interesting: There is indeed a delay at the start (720 samples in a > quick test) compared to the native AAC decoder. > Furthermore, the current code is buggy, as it believes that > avcodec->time_base to be the time_base of the returned AVFrames (it is > in reality avcodec->pkt_timebase; just test with AAC-in-Matroska for this). > I haven't tested your patches, but I have now realized that there is > indeed an issue. And your patch should also fix the wrong timebase issue. Based on discussion on irc with Andreas and James, I think the conclusion was that this patch should be fine, so I'll go ahead and push it soon if there's no further comments on it. // Martin _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 6+ messages in thread
end of thread, other threads:[~2022-02-03 11:53 UTC | newest] Thread overview: 6+ messages (download: mbox.gz / follow: Atom feed) -- links below jump to the message on this page -- 2022-01-19 20:38 [FFmpeg-devel] [PATCH 1/2] libfdk-aacdec: Add an option for setting the decoder's DRC album mode Martin Storsjö 2022-01-19 20:38 ` [FFmpeg-devel] [PATCH 2/2] libfdk-aacdec: Flush delayed samples at the end Martin Storsjö 2022-01-21 13:00 ` Andreas Rheinhardt 2022-01-21 13:58 ` Martin Storsjö 2022-01-21 14:42 ` Andreas Rheinhardt 2022-02-03 11:53 ` Martin Storsjö
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