From c96bd21227e594856f8fd0610fd213b002056383 Mon Sep 17 00:00:00 2001 From: Michael Niedermayer Date: Sun, 21 Dec 2008 23:50:16 +0000 Subject: [PATCH] Change mpeg audio parser so it only sets frame_size, channels and bit_rate after it has a few valid frames. Fixes issue762 Originally committed as revision 16259 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/mpegaudio.h | 2 +- libavcodec/mpegaudio_parser.c | 24 ++++++++++++++---------- libavformat/mp3.c | 2 +- 3 files changed, 16 insertions(+), 12 deletions(-) diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 21de2da..33af18d 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -128,7 +128,7 @@ typedef struct HuffTable { } HuffTable; int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf); -int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate); +int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate); void ff_mpa_synth_init(MPA_INT *window); void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, MPA_INT *window, int *dither_state, diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c index e7cb743..e4b17b0 100644 --- a/libavcodec/mpegaudio_parser.c +++ b/libavcodec/mpegaudio_parser.c @@ -44,7 +44,7 @@ typedef struct MpegAudioParseContext { /* useful helper to get mpeg audio stream infos. Return -1 if error in header, otherwise the coded frame size in bytes */ -int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate) +int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate) { MPADecodeContext s1, *s = &s1; s1.avctx = avctx; @@ -58,23 +58,23 @@ int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate) switch(s->layer) { case 1: - avctx->frame_size = 384; + *frame_size = 384; break; case 2: - avctx->frame_size = 1152; + *frame_size = 1152; break; default: case 3: if (s->lsf) - avctx->frame_size = 576; + *frame_size = 576; else - avctx->frame_size = 1152; + *frame_size = 1152; break; } *sample_rate = s->sample_rate; - avctx->channels = s->nb_channels; - avctx->bit_rate = s->bit_rate; + *channels = s->nb_channels; + *bit_rate = s->bit_rate; avctx->sub_id = s->layer; return s->frame_size; } @@ -92,7 +92,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1, const uint8_t *buf, int buf_size) { MpegAudioParseContext *s = s1->priv_data; - int len, ret, sr; + int len, ret, sr, channels, bit_rate, frame_size; uint32_t header; const uint8_t *buf_ptr; @@ -123,7 +123,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1, got_header: header = AV_RB32(s->inbuf); - ret = ff_mpa_decode_header(avctx, header, &sr); + ret = ff_mpa_decode_header(avctx, header, &sr, &channels, &frame_size, &bit_rate); if (ret < 0) { s->header_count= -2; /* no sync found : move by one byte (inefficient, but simple!) */ @@ -146,8 +146,12 @@ static int mpegaudio_parse(AVCodecParserContext *s1, s->frame_size = -1; } #endif - if(s->header_count > 1) + if(s->header_count > 1){ avctx->sample_rate= sr; + avctx->channels = channels; + avctx->frame_size = frame_size; + avctx->bit_rate = bit_rate; + } } } } else diff --git a/libavformat/mp3.c b/libavformat/mp3.c index 407d4f0..0f3a5fe 100644 --- a/libavformat/mp3.c +++ b/libavformat/mp3.c @@ -383,7 +383,7 @@ static int mp3_read_probe(AVProbeData *p) for(frames = 0; buf2 < end; frames++) { header = AV_RB32(buf2); - fsize = ff_mpa_decode_header(&avctx, header, &sample_rate); + fsize = ff_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate); if(fsize < 0) break; buf2 += fsize; -- 2.7.4