generic-library/ffmpeg
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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  mov: Don't av_malloc(0).
  avconv: only allocate 1 AVFrame per input stream
  avconv: fix memleaks due to not freeing the AVFrame for audio
  h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
  misc Doxygen markup improvements
  doxygen: eliminate Qt-style doxygen syntax
  g722: Add a regression test for muxing/demuxing in wav
  g722: Change bits per sample to 4
  g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
  api-example: update to use avcodec_decode_audio4()
  avplay: use avcodec_decode_audio4()
  avplay: use a separate buffer for playing silence
  avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
  avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
  mov: Allow empty stts atom.
  doc: document preferred Doxygen syntax and make patcheck detect it

Conflicts:
	avconv.c
	ffplay.c
	libavcodec/mlpdec.c
	libavcodec/version.h
	libavformat/mov.c
	tests/codec-regression.sh
	tests/fate/h264.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-12-06 01:37:27 +01:00
parent a448a5d1c4 52401b82bd
commit b404ab9e74
34 changed files with 330 additions and 272 deletions
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85
ffmpeg.c
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@@ -168,7 +168,6 @@ static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
static void *samples;
static uint8_t *input_tmp= NULL;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
@@ -179,6 +178,8 @@ typedef struct InputStream {
int discard; /* true if stream data should be discarded */
int decoding_needed; /* true if the packets must be decoded in 'raw_fifo' */
AVCodec *dec;
AVFrame *decoded_frame;
AVFrame *filtered_frame;
int64_t start; /* time when read started */
int64_t next_pts; /* synthetic pts for cases where pkt.pts
@@ -658,8 +659,11 @@ void av_noreturn exit_program(int ret)
for(i=0;i<nb_input_files;i++) {
av_close_input_file(input_files[i].ctx);
}
for (i = 0; i < nb_input_streams; i++)
for (i = 0; i < nb_input_streams; i++) {
av_freep(&input_streams[i].decoded_frame);
av_freep(&input_streams[i].filtered_frame);
av_dict_free(&input_streams[i].opts);
}
if (vstats_file)
fclose(vstats_file);
@@ -674,7 +678,6 @@ void av_noreturn exit_program(int ret)
av_free(audio_buf);
av_free(audio_out);
allocated_audio_buf_size= allocated_audio_out_size= 0;
av_free(samples);
#if CONFIG_AVFILTER
avfilter_uninit();
@@ -838,14 +841,11 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
memset(buf, fill_char, size);
}
static void do_audio_out(AVFormatContext *s,
OutputStream *ost,
InputStream *ist,
unsigned char *buf, int size)
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
InputStream *ist, AVFrame *decoded_frame)
{
uint8_t *buftmp;
int64_t audio_out_size, audio_buf_size;
int64_t allocated_for_size= size;
int size_out, frame_bytes, ret, resample_changed;
AVCodecContext *enc= ost->st->codec;
@@ -853,6 +853,9 @@ static void do_audio_out(AVFormatContext *s,
int osize = av_get_bytes_per_sample(enc->sample_fmt);
int isize = av_get_bytes_per_sample(dec->sample_fmt);
const int coded_bps = av_get_bits_per_sample(enc->codec->id);
uint8_t *buf = decoded_frame->data[0];
int size = decoded_frame->nb_samples * dec->channels * isize;
int64_t allocated_for_size = size;
need_realloc:
audio_buf_size= (allocated_for_size + isize*dec->channels - 1) / (isize*dec->channels);
@@ -1732,39 +1735,42 @@ static void rate_emu_sleep(InputStream *ist)
static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
{
static unsigned int samples_size = 0;
AVFrame *decoded_frame;
AVCodecContext *avctx = ist->st->codec;
int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
uint8_t *decoded_data_buf = NULL;
int decoded_data_size = 0;
int i, ret;
if (pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
av_free(samples);
samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
samples = av_malloc(samples_size);
}
decoded_data_size = samples_size;
if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
else
avcodec_get_frame_defaults(ist->decoded_frame);
decoded_frame = ist->decoded_frame;
ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
pkt);
if (ret < 0)
ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt);
if (ret < 0) {
return ret;
*got_output = decoded_data_size > 0;
}
/* Some bug in mpeg audio decoder gives */
/* decoded_data_size < 0, it seems they are overflows */
if (!*got_output) {
/* no audio frame */
return ret;
}
decoded_data_buf = (uint8_t *)samples;
ist->next_pts += ((int64_t)AV_TIME_BASE/bps * decoded_data_size) /
(ist->st->codec->sample_rate * ist->st->codec->channels);
/* if the decoder provides a pts, use it instead of the last packet pts.
the decoder could be delaying output by a packet or more. */
if (decoded_frame->pts != AV_NOPTS_VALUE)
ist->next_pts = decoded_frame->pts;
/* increment next_pts to use for the case where the input stream does not
have timestamps or there are multiple frames in the packet */
ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
avctx->sample_rate;
// preprocess audio (volume)
if (audio_volume != 256) {
switch (ist->st->codec->sample_fmt) {
int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;
void *samples = decoded_frame->data[0];
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *volp = samples;
@@ -1825,9 +1831,9 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
continue;
do_audio_out(output_files[ost->file_index].ctx, ost, ist,
decoded_data_buf, decoded_data_size);
do_audio_out(output_files[ost->file_index].ctx, ost, ist, decoded_frame);
}
return ret;
}
@@ -1844,8 +1850,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
int64_t *best_effort_timestamp;
AVRational *frame_sample_aspect;
if (!(decoded_frame = avcodec_alloc_frame()))
if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
else
avcodec_get_frame_defaults(ist->decoded_frame);
decoded_frame = ist->decoded_frame;
pkt->pts = *pkt_pts;
pkt->dts = *pkt_dts;
*pkt_pts = AV_NOPTS_VALUE;
@@ -1867,12 +1876,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
ret = avcodec_decode_video2(ist->st->codec,
decoded_frame, got_output, pkt);
if (ret < 0)
goto fail;
return ret;
quality = same_quant ? decoded_frame->quality : 0;
if (!*got_output) {
/* no picture yet */
av_freep(&decoded_frame);
return ret;
}
@@ -1917,10 +1925,12 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
AVRational ist_pts_tb = ost->output_video_filter->inputs[0]->time_base;
if (av_buffersink_get_buffer_ref(ost->output_video_filter, &ost->picref, 0) < 0)
goto cont;
if (!filtered_frame && !(filtered_frame = avcodec_alloc_frame())) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (!ist->filtered_frame && !(ist->filtered_frame = avcodec_alloc_frame())) {
av_free(buffer_to_free);
return AVERROR(ENOMEM);
} else
avcodec_get_frame_defaults(ist->filtered_frame);
filtered_frame = ist->filtered_frame;
*filtered_frame= *decoded_frame; //for me_threshold
if (ost->picref) {
avfilter_fill_frame_from_video_buffer_ref(filtered_frame, ost->picref);
@@ -1942,13 +1952,10 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
frame_available = ost->output_video_filter && avfilter_poll_frame(ost->output_video_filter->inputs[0]);
avfilter_unref_buffer(ost->picref);
}
av_freep(&filtered_frame);
#endif
}
fail:
av_free(buffer_to_free);
av_freep(&decoded_frame);
return ret;
}