Commit c40a35f8 authored by Michael Niedermayer's avatar Michael Niedermayer

Merge remote-tracking branch 'qatar/master'

* qatar/master:
  indeo3: check per-plane data buffer against input buffer bounds.
  avconv: Handle audio sync for non-S16 sample formats.
  pthread: don't increment index on zero-sized packets.

Conflicts:
	libavcodec/indeo3.c
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents 23b4f355 464ccb01
......@@ -779,6 +779,14 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx
}
}
static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static void do_audio_out(AVFormatContext *s,
OutputStream *ost,
InputStream *ist,
......@@ -879,9 +887,9 @@ need_realloc:
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
- av_fifo_size(ost->fifo)/(enc->channels * 2);
double idelta= delta*dec->sample_rate / enc->sample_rate;
int byte_delta= ((int)idelta)*2*dec->channels;
- av_fifo_size(ost->fifo)/(enc->channels * osize);
int idelta = delta * dec->sample_rate / enc->sample_rate;
int byte_delta = idelta * isize * dec->channels;
//FIXME resample delay
if(fabs(delta) > 50){
......@@ -890,7 +898,8 @@ need_realloc:
byte_delta= FFMAX(byte_delta, -size);
size += byte_delta;
buf -= byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n", (int)-delta);
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n",
-byte_delta / (isize * dec->channels));
if(!size)
return;
ist->is_start=0;
......@@ -904,11 +913,11 @@ need_realloc:
}
ist->is_start=0;
memset(input_tmp, 0, byte_delta);
generate_silence(input_tmp, dec->sample_fmt, byte_delta);
memcpy(input_tmp + byte_delta, buf, size);
buf= input_tmp;
size += byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", (int)delta);
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
}
}else if(audio_sync_method>1){
int comp= av_clip(delta, -audio_sync_method, audio_sync_method);
......@@ -921,7 +930,7 @@ need_realloc:
}
}else
ost->sync_opts= lrintf(get_sync_ipts(ost) * enc->sample_rate)
- av_fifo_size(ost->fifo)/(enc->channels * 2); //FIXME wrong
- av_fifo_size(ost->fifo)/(enc->channels * osize); //FIXME wrong
if (ost->audio_resample) {
buftmp = audio_buf;
......@@ -1505,14 +1514,6 @@ static void print_report(OutputFile *output_files,
}
}
static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
{
int i, ret;
......
......@@ -827,6 +827,14 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx
}
}
static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static void do_audio_out(AVFormatContext *s,
OutputStream *ost,
InputStream *ist,
......@@ -926,9 +934,9 @@ need_realloc:
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
- av_fifo_size(ost->fifo)/(enc->channels * 2);
double idelta= delta*dec->sample_rate / enc->sample_rate;
int byte_delta= ((int)idelta)*2*dec->channels;
- av_fifo_size(ost->fifo)/(enc->channels * osize);
int idelta = delta * dec->sample_rate / enc->sample_rate;
int byte_delta = idelta * isize * dec->channels;
//FIXME resample delay
if(fabs(delta) > 50){
......@@ -937,7 +945,8 @@ need_realloc:
byte_delta= FFMAX(byte_delta, -size);
size += byte_delta;
buf -= byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n", (int)-delta);
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n",
-byte_delta / (isize * dec->channels));
if(!size)
return;
ist->is_start=0;
......@@ -950,11 +959,11 @@ need_realloc:
}
ist->is_start=0;
memset(input_tmp, 0, byte_delta);
generate_silence(input_tmp, dec->sample_fmt, byte_delta);
memcpy(input_tmp + byte_delta, buf, size);
buf= input_tmp;
size += byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", (int)delta);
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
}
}else if(audio_sync_method>1){
int comp= av_clip(delta, -audio_sync_method, audio_sync_method);
......@@ -967,7 +976,7 @@ need_realloc:
}
}else
ost->sync_opts= lrintf(get_sync_ipts(ost) * enc->sample_rate)
- av_fifo_size(ost->fifo)/(enc->channels * 2); //FIXME wrong
- av_fifo_size(ost->fifo)/(enc->channels * osize); //FIXME wrong
if (ost->audio_resample) {
buftmp = audio_buf;
......@@ -1535,14 +1544,6 @@ static void print_report(OutputFile *output_files,
}
}
static void generate_silence(uint8_t *buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
{
int i, ret;
......
......@@ -494,6 +494,7 @@ static int submit_packet(PerThreadContext *p, AVPacket *avpkt)
}
fctx->prev_thread = p;
fctx->next_decoding++;
return 0;
}
......@@ -516,8 +517,6 @@ int ff_thread_decode_frame(AVCodecContext *avctx,
err = submit_packet(p, avpkt);
if (err) return err;
fctx->next_decoding++;
/*
* If we're still receiving the initial packets, don't return a frame.
*/
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment