opensource/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2025-08-06 15:49:50 +08:00
Code Issues Packages Projects Releases Wiki Activity

Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  rv34: Handle only complete frames in frame-mt.
  MPV: set reference frame pointers to NULL when allocation of dummy pictures fails
  configure: die if x11grab dependencies are unavailable
  zerocodec: factorize loop
  avconv: fix the resampling safety factors for output audio buffer allocation
  avconv: move audio output buffer allocation to a separate function
  avconv: make the async buffer global and free it in exit_program()

Conflicts:
	ffmpeg.c
	libavcodec/mpegvideo.c
	libavcodec/rv34.c
	libavcodec/zerocodec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-03-23 21:20:06 +01:00
parent adb98a3d22 73ad4471a4
commit e2d110d8d2
5 changed files with 175 additions and 131 deletions
Download Patch File Download Diff File Expand all files Collapse all files

74
ffmpeg.c
View File

@ -179,8 +179,8 @@ static int debug_ts = 0;
static uint8_t *audio_buf;
static unsigned int allocated_audio_buf_size;
static uint8_t *input_tmp= NULL;
static uint8_t *async_buf;
static unsigned int allocated_async_buf_size;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
@ -868,16 +868,16 @@ void av_noreturn exit_program(int ret)
av_freep(&output_files);
uninit_opts();
av_free(audio_buf);
av_freep(&audio_buf);
allocated_audio_buf_size = 0;
av_freep(&async_buf);
allocated_async_buf_size = 0;
#if CONFIG_AVFILTER
avfilter_uninit();
#endif
avformat_network_deinit();
av_freep(&input_tmp);
if (received_sigterm) {
av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
(int) received_sigterm);
@ -1126,11 +1126,38 @@ static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
return ret;
}
static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
int nb_samples)
{
int64_t audio_buf_samples;
int audio_buf_size;
/* calculate required number of samples to allocate */
audio_buf_samples = ((int64_t)nb_samples * enc->sample_rate + dec->sample_rate) /
dec->sample_rate;
audio_buf_samples = 4 * audio_buf_samples + 10000; // safety factors for resampling
audio_buf_samples = FFMAX(audio_buf_samples, enc->frame_size);
if (audio_buf_samples > INT_MAX)
return AVERROR(EINVAL);
audio_buf_size = av_samples_get_buffer_size(NULL, enc->channels,
audio_buf_samples,
enc->sample_fmt, 32);
if (audio_buf_size < 0)
return audio_buf_size;
av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
if (!audio_buf)
return AVERROR(ENOMEM);
return 0;
}
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
InputStream *ist, AVFrame *decoded_frame)
{
uint8_t *buftmp;
int64_t audio_buf_size, size_out;
int64_t size_out;
int frame_bytes, resample_changed;
AVCodecContext *enc = ost->st->codec;
@ -1140,7 +1167,6 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
uint8_t *buf[AV_NUM_DATA_POINTERS];
int size = decoded_frame->nb_samples * dec->channels * isize;
int planes = av_sample_fmt_is_planar(dec->sample_fmt) ? dec->channels : 1;
int64_t allocated_for_size = size;
int i;
av_assert0(planes <= AV_NUM_DATA_POINTERS);
@ -1148,21 +1174,8 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
for(i=0; i<planes; i++)
buf[i]= decoded_frame->data[i];
need_realloc:
audio_buf_size = (allocated_for_size + isize * dec->channels - 1) / (isize * dec->channels);
audio_buf_size = (audio_buf_size * enc->sample_rate + dec->sample_rate) / dec->sample_rate;
audio_buf_size = audio_buf_size * 2 + 10000; // safety factors for the deprecated resampling API
audio_buf_size = FFMAX(audio_buf_size, enc->frame_size);
audio_buf_size *= osize * enc->channels;
if (audio_buf_size > INT_MAX) {
av_log(NULL, AV_LOG_FATAL, "Buffer sizes too large\n");
exit_program(1);
}
av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
if (!audio_buf) {
av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
@ -1251,16 +1264,21 @@ need_realloc:
return;
ist->is_start = 0;
} else {
input_tmp = av_realloc(input_tmp, byte_delta + size);
av_fast_malloc(&async_buf, &allocated_async_buf_size,
byte_delta + size);
if (!async_buf) {
av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
exit_program(1);
}
if (byte_delta > allocated_for_size - size) {
allocated_for_size = byte_delta + (int64_t)size;
goto need_realloc;
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
ist->is_start = 0;
for (i=0; i<planes; i++) {
uint8_t *t = input_tmp + i*((byte_delta + size)/planes);
uint8_t *t = async_buf + i*((byte_delta + size)/planes);
generate_silence(t, dec->sample_fmt, byte_delta/planes);
memcpy(t + byte_delta/planes, buf[i], size/planes);
buf[i] = t;
@ -1283,7 +1301,7 @@ need_realloc:
if (ost->audio_resample || ost->audio_channels_mapped) {
buftmp = audio_buf;
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, audio_buf_size / (enc->channels * osize),
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, allocated_audio_buf_size / (enc->channels * osize),
buf, size / (dec->channels * isize));
if (size_out < 0) {
av_log(NULL, AV_LOG_FATAL, "swr_convert failed\n");