From aef2016bb02fba377481789bf6a84e1176b83c25 Mon Sep 17 00:00:00 2001 From: Marton Balint Date: Fri, 28 Feb 2020 00:26:20 +0100 Subject: avformat/audiointerleave: disallow using a samples_per_frame array Only MXF used an actual sample array, and that is unneeded there because simple rounding rules can be used instead. Signed-off-by: Marton Balint --- libavformat/audiointerleave.c | 24 +++++++++++------------- libavformat/audiointerleave.h | 7 ++++--- libavformat/gxfenc.c | 2 +- libavformat/mxfenc.c | 7 ++----- 4 files changed, 18 insertions(+), 22 deletions(-) diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c index 6797546a44..2e83031bd6 100644 --- a/libavformat/audiointerleave.c +++ b/libavformat/audiointerleave.c @@ -39,14 +39,11 @@ void ff_audio_interleave_close(AVFormatContext *s) } int ff_audio_interleave_init(AVFormatContext *s, - const int *samples_per_frame, + const int samples_per_frame, AVRational time_base) { int i; - if (!samples_per_frame) - return AVERROR(EINVAL); - if (!time_base.num) { av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); return AVERROR(EINVAL); @@ -56,6 +53,8 @@ int ff_audio_interleave_init(AVFormatContext *s, AudioInterleaveContext *aic = st->priv_data; if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { + int max_samples = samples_per_frame ? samples_per_frame : + av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP); aic->sample_size = (st->codecpar->channels * av_get_bits_per_sample(st->codecpar->codec_id)) / 8; if (!aic->sample_size) { @@ -63,12 +62,11 @@ int ff_audio_interleave_init(AVFormatContext *s, return AVERROR(EINVAL); } aic->samples_per_frame = samples_per_frame; - aic->samples = aic->samples_per_frame; aic->time_base = time_base; - aic->fifo_size = 100* *aic->samples; - if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples))) + if (!(aic->fifo = av_fifo_alloc_array(100, max_samples))) return AVERROR(ENOMEM); + aic->fifo_size = 100 * max_samples; } } @@ -81,7 +79,9 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st = s->streams[stream_index]; AudioInterleaveContext *aic = st->priv_data; int ret; - int frame_size = *aic->samples * aic->sample_size; + int nb_samples = aic->samples_per_frame ? aic->samples_per_frame : + (av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples); + int frame_size = nb_samples * aic->sample_size; int size = FFMIN(av_fifo_size(aic->fifo), frame_size); if (!size || (!flush && size == av_fifo_size(aic->fifo))) return 0; @@ -95,13 +95,11 @@ static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, memset(pkt->data + size, 0, pkt->size - size); pkt->dts = pkt->pts = aic->dts; - pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); + pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base); pkt->stream_index = stream_index; aic->dts += pkt->duration; - - aic->samples++; - if (!*aic->samples) - aic->samples = aic->samples_per_frame; + aic->nb_samples += nb_samples; + aic->n++; return pkt->size; } diff --git a/libavformat/audiointerleave.h b/libavformat/audiointerleave.h index f28d5fefcc..0933310f4c 100644 --- a/libavformat/audiointerleave.h +++ b/libavformat/audiointerleave.h @@ -29,14 +29,15 @@ typedef struct AudioInterleaveContext { AVFifoBuffer *fifo; unsigned fifo_size; ///< size of currently allocated FIFO + int64_t n; ///< number of generated packets + int64_t nb_samples; ///< number of generated samples uint64_t dts; ///< current dts int sample_size; ///< size of one sample all channels included - const int *samples_per_frame; ///< must be 0-terminated - const int *samples; ///< current samples per frame, pointer to samples_per_frame + int samples_per_frame; ///< samples per frame if fixed, 0 otherwise AVRational time_base; ///< time base of output audio packets } AudioInterleaveContext; -int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base); +int ff_audio_interleave_init(AVFormatContext *s, const int samples_per_frame, AVRational time_base); void ff_audio_interleave_close(AVFormatContext *s); /** diff --git a/libavformat/gxfenc.c b/libavformat/gxfenc.c index 9eebefc683..e7536a6a7e 100644 --- a/libavformat/gxfenc.c +++ b/libavformat/gxfenc.c @@ -663,7 +663,7 @@ static int gxf_write_umf_packet(AVFormatContext *s) return updatePacketSize(pb, pos); } -static const int GXF_samples_per_frame[] = { 32768, 0 }; +static const int GXF_samples_per_frame = 32768; static void gxf_init_timecode_track(GXFStreamContext *sc, GXFStreamContext *vsc) { diff --git a/libavformat/mxfenc.c b/libavformat/mxfenc.c index 55c715d776..cbb4d9cc9a 100644 --- a/libavformat/mxfenc.c +++ b/libavformat/mxfenc.c @@ -1747,7 +1747,7 @@ static void mxf_write_index_table_segment(AVFormatContext *s) avio_wb32(pb, KAG_SIZE); // system item size including klv fill } else { // audio or data track if (!audio_frame_size) { - audio_frame_size = sc->aic.samples[0]*sc->aic.sample_size; + audio_frame_size = sc->frame_size; audio_frame_size += klv_fill_size(audio_frame_size); } avio_w8(pb, 1); @@ -2650,10 +2650,7 @@ static int mxf_write_header(AVFormatContext *s) return AVERROR(ENOMEM); mxf->timecode_track->index = -1; - if (!spf) - spf = ff_mxf_get_samples_per_frame(s, (AVRational){ 1, 25 }); - - if (ff_audio_interleave_init(s, spf->samples_per_frame, mxf->time_base) < 0) + if (ff_audio_interleave_init(s, 0, av_inv_q(mxf->tc.rate)) < 0) return -1; return 0; -- cgit v1.2.3