generic-library/ffmpeg
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Fix encoding when the input audio format/rate/channels changes during

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transcoding.

Fix issue #2292.

Patch sponsored by KIM Keep In Mind GmbH, srl.

Originally committed as revision 25939 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Stefano Sabatini 2010-12-12 16:18:50 +00:00
parent 4ba22e044b
commit 8afab68605
1 changed files with 30 additions and 2 deletions
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32
ffmpeg.c
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@ -295,6 +295,9 @@ typedef struct AVOutputStream {
/* audio only */ /* audio only */
int audio_resample; int audio_resample;
ReSampleContext *resample; /* for audio resampling */ ReSampleContext *resample; /* for audio resampling */
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
int reformat_pair; int reformat_pair;
AVAudioConvert *reformat_ctx; AVAudioConvert *reformat_ctx;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */ AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
@ -768,7 +771,7 @@ static void do_audio_out(AVFormatContext *s,
int64_t audio_out_size, audio_buf_size; int64_t audio_out_size, audio_buf_size;
int64_t allocated_for_size= size; int64_t allocated_for_size= size;
int size_out, frame_bytes, ret; int size_out, frame_bytes, ret, resample_changed;
AVCodecContext *enc= ost->st->codec; AVCodecContext *enc= ost->st->codec;
AVCodecContext *dec= ist->st->codec; AVCodecContext *dec= ist->st->codec;
int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8; int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;
@ -802,7 +805,28 @@ need_realloc:
if (enc->channels != dec->channels) if (enc->channels != dec->channels)
ost->audio_resample = 1; ost->audio_resample = 1;
if (ost->audio_resample && !ost->resample) { resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->resample) || resample_changed) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d.%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
ist->file_index, ist->index,
ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_sample_rate = dec->sample_rate;
if (ost->resample)
audio_resample_close(ost->resample);
}
if (ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
ost->resample = NULL;
ost->audio_resample = 0;
} else {
if (dec->sample_fmt != AV_SAMPLE_FMT_S16) if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n"); fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels, ost->resample = av_audio_resample_init(enc->channels, dec->channels,
@ -815,6 +839,7 @@ need_realloc:
enc->channels, enc->sample_rate); enc->channels, enc->sample_rate);
ffmpeg_exit(1); ffmpeg_exit(1);
} }
}
} }
#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b)) #define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
@ -2174,6 +2199,9 @@ static int transcode(AVFormatContext **output_files,
icodec->request_channels = codec->channels; icodec->request_channels = codec->channels;
ist->decoding_needed = 1; ist->decoding_needed = 1;
ost->encoding_needed = 1; ost->encoding_needed = 1;
ost->resample_sample_fmt = icodec->sample_fmt;
ost->resample_sample_rate = icodec->sample_rate;
ost->resample_channels = icodec->channels;
break; break;
case AVMEDIA_TYPE_VIDEO: case AVMEDIA_TYPE_VIDEO:
if (ost->st->codec->pix_fmt == PIX_FMT_NONE) { if (ost->st->codec->pix_fmt == PIX_FMT_NONE) {