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ffmpeg.wasm-core
Commit b404ab9e authored by Michael Niedermayer's avatar Michael Niedermayer
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Merge remote-tracking branch 'qatar/master' 

* 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: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents a448a5d1 52401b82
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Showing with 330 additions and 272 deletions
avconv.c
View file @ b404ab9e
......@@ -155,8 +155,6 @@ static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
static void *samples;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
typedef struct InputStream {
......@@ -165,6 +163,8 @@ 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
......@@ -612,8 +612,11 @@ void 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);
......@@ -628,7 +631,6 @@ void 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();
......@@ -787,14 +789,11 @@ 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;
......@@ -802,6 +801,9 @@ 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);
......@@ -1697,39 +1699,42 @@ 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;
......@@ -1790,9 +1795,9 @@ 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;
}
......@@ -1806,8 +1811,11 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
int frame_available = 1;
#endif
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 = ist->pts;
*pkt_pts = AV_NOPTS_VALUE;
......@@ -1815,12 +1823,11 @@ 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;
}
ist->next_pts = ist->pts = decoded_frame->best_effort_timestamp;
......@@ -1852,10 +1859,12 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
decoded_frame->pts = ist->pts;
av_vsrc_buffer_add_frame(ost->input_video_filter, decoded_frame, AV_VSRC_BUF_FLAG_OVERWRITE);
if (!(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;
frame_available = avfilter_poll_frame(ost->output_video_filter->inputs[0]);
}
while (frame_available) {
......@@ -1884,13 +1893,10 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
if (ost->picref)
avfilter_unref_buffer(ost->picref);
}
av_freep(&filtered_frame);
#endif
}
fail:
av_free(buffer_to_free);
av_freep(&decoded_frame);
return ret;
}
......
cmdutils.h
View file @ b404ab9e
......@@ -196,7 +196,7 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options);
*
* @param s Corresponding format context.
* @param st Stream from s to be checked.
* @param spec A stream specifier of the [v|a|s|d]:[<stream index>] form.
* @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
*
* @return 1 if the stream matches, 0 if it doesn't, <0 on error
*/
......
doc/developer.texi
View file @ b404ab9e
......@@ -77,6 +77,11 @@ Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
@code{//!} with @code{///} and similar. Also @@ syntax should be employed
for markup commands, i.e. use @code{@@param} and not @code{\param}.
@example
/**
* @@file
......
doc/examples/decoding_encoding.c
View file @ b404ab9e
......@@ -33,6 +33,7 @@
#include "libavutil/opt.h"
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
......@@ -114,11 +115,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int out_size, len;
int len;
FILE *f, *outfile;
uint8_t *outbuf;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
......@@ -139,8 +140,6 @@ static void audio_decode_example(const char *outfilename, const char *filename)
exit(1);
}
outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
......@@ -157,15 +156,27 @@ static void audio_decode_example(const char *outfilename, const char *filename)
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = avcodec_alloc_frame())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
} else
avcodec_get_frame_defaults(decoded_frame);
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (out_size > 0) {
if (got_frame) {
/* if a frame has been decoded, output it */
fwrite(outbuf, 1, out_size, outfile);
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
......@@ -185,10 +196,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
fclose(outfile);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
av_free(decoded_frame);
}
/*
......
ffmpeg.c
View file @ b404ab9e
......@@ -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;
}
......
ffplay.c
View file @ b404ab9e
......@@ -151,11 +151,10 @@ typedef struct VideoState {
AVStream *audio_st;
PacketQueue audioq;
int audio_hw_buf_size;
/* samples output by the codec. we reserve more space for avsync
compensation, resampling and format conversion */
DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
uint8_t *audio_buf;
uint8_t *audio_buf1;
unsigned int audio_buf_size; /* in bytes */
int audio_buf_index; /* in bytes */
int audio_write_buf_size;
......@@ -174,6 +173,7 @@ typedef struct VideoState {
double audio_current_pts_drift;
int frame_drops_early;
int frame_drops_late;
AVFrame *frame;
enum ShowMode {
SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
......@@ -1998,8 +1998,8 @@ static int synchronize_audio(VideoState *is, short *samples,
max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
if (wanted_size < min_size)
wanted_size = min_size;
else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)))
wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2));
else if (wanted_size > FFMIN3(max_size, samples_size, sizeof(is->audio_buf2)))
wanted_size = FFMIN3(max_size, samples_size, sizeof(is->audio_buf2));
/* add or remove samples to correction the synchro */
if (wanted_size < samples_size) {
......@@ -2043,7 +2043,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
AVPacket *pkt = &is->audio_pkt;
AVCodecContext *dec= is->audio_st->codec;
int len1, len2, data_size, resampled_data_size;
int64_t dec_channel_layout;
int64_t dec_channel_layout, got_frame;
double pts;
int new_packet = 0;
int flush_complete = 0;
......@@ -2051,13 +2051,16 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
for(;;) {
/* NOTE: the audio packet can contain several frames */
while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
if (!is->frame) {
if (!(is->frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
} else
avcodec_get_frame_defaults(is->frame);
if (flush_complete)
break;
new_packet = 0;
data_size = sizeof(is->audio_buf1);
len1 = avcodec_decode_audio3(dec,
(int16_t *)is->audio_buf1, &data_size,
pkt_temp);
len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
if (len1 < 0) {
/* if error, we skip the frame */
pkt_temp->size = 0;
......@@ -2067,12 +2070,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
pkt_temp->data += len1;
pkt_temp->size -= len1;
if (data_size <= 0) {
if (!got_frame) {
/* stop sending empty packets if the decoder is finished */
if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
flush_complete = 1;
continue;
}
data_size = av_samples_get_buffer_size(NULL, dec->channels,
is->frame->nb_samples,
dec->sample_fmt, 1);
dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
......@@ -2101,7 +2107,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
resampled_data_size = data_size;
if (is->swr_ctx) {
const uint8_t *in[] = {is->audio_buf1};
const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2};
len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
......@@ -2116,7 +2122,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
is->audio_buf = is->audio_buf2;
resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
} else {
is->audio_buf= is->audio_buf1;
is->audio_buf = is->frame->data[0];
}
/* if no pts, then compute it */
......@@ -2150,11 +2156,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (pkt->data == flush_pkt.data)
avcodec_flush_buffers(dec);
pkt_temp->data = pkt->data;
pkt_temp->size = pkt->size;
pkt_temp->flags = pkt->flags;
pkt_temp->side_data = pkt->side_data;
pkt_temp->side_data_elems = pkt->side_data_elems;
*pkt_temp = *pkt;
/* if update the audio clock with the pts */
if (pkt->pts != AV_NOPTS_VALUE) {
......@@ -2178,9 +2180,8 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
audio_size = audio_decode_frame(is, &pts);
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf = is->audio_buf1;
is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
memset(is->audio_buf, 0, is->audio_buf_size);
is->audio_buf = is->silence_buf;
is->audio_buf_size = sizeof(is->silence_buf);
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
......@@ -2356,6 +2357,9 @@ static void stream_component_close(VideoState *is, int stream_index)
if (is->swr_ctx)
swr_free(&is->swr_ctx);
av_free_packet(&is->audio_pkt);
av_freep(&is->audio_buf1);
is->audio_buf = NULL;
av_freep(&is->frame);
if (is->rdft) {
av_rdft_end(is->rdft);
......
libavcodec/amrnbdec.c
View file @ b404ab9e
......@@ -653,7 +653,7 @@ static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
static void apply_ir_filter(float *out, const AMRFixed *in,
const float *filter)
{
float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2
float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
filter2[AMR_SUBFRAME_SIZE];
int lag = in->pitch_lag;
float fac = in->pitch_fac;
......
libavcodec/cinepak.c
View file @ b404ab9e
......@@ -22,10 +22,11 @@
/**
* @file
* Cinepak video decoder
* by Ewald Snel <ewald@rambo.its.tudelft.nl>
* For more information on the Cinepak algorithm, visit:
* @author Ewald Snel <ewald@rambo.its.tudelft.nl>
*
* @see For more information on the Cinepak algorithm, visit:
* http://www.csse.monash.edu.au/~timf/
* For more information on the quirky data inside Sega FILM/CPK files, visit:
* @see For more information on the quirky data inside Sega FILM/CPK files, visit:
* http://wiki.multimedia.cx/index.php?title=Sega_FILM
*/
......
libavcodec/eamad.c
View file @ b404ab9e
......@@ -22,9 +22,9 @@
/**
* @file
* Electronic Arts Madcow Video Decoder
* by Peter Ross <pross@xvid.org>
* @author Peter Ross <pross@xvid.org>
*
* Technical details here:
* @see technical details at
* http://wiki.multimedia.cx/index.php?title=Electronic_Arts_MAD
*/
......
libavcodec/g722.h
View file @ b404ab9e
......@@ -31,7 +31,9 @@
#define PREV_SAMPLES_BUF_SIZE 1024
typedef struct {
const AVClass *class;
AVFrame frame;
int bits_per_codeword;
int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
int prev_samples_pos; ///< the number of values in prev_samples
......
libavcodec/g722dec.c
View file @ b404ab9e
......@@ -37,6 +37,21 @@
#include "avcodec.h"
#include "get_bits.h"
#include "g722.h"
#include "libavutil/opt.h"
#define OFFSET(x) offsetof(G722Context, x)
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "bits_per_codeword", "Bits per G722 codeword", OFFSET(bits_per_codeword), AV_OPT_TYPE_FLAGS, { 8 }, 6, 8, AD },
{ NULL }
};
static const AVClass g722_decoder_class = {
.class_name = "g722 decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static av_cold int g722_decode_init(AVCodecContext * avctx)
{
......@@ -48,20 +63,6 @@ static av_cold int g722_decode_init(AVCodecContext * avctx)
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
switch (avctx->bits_per_coded_sample) {
case 8:
case 7:
case 6:
break;
default:
av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], "
"assuming 8\n",
avctx->bits_per_coded_sample);
case 0:
avctx->bits_per_coded_sample = 8;
break;
}
c->band[0].scale_factor = 8;
c->band[1].scale_factor = 2;
c->prev_samples_pos = 22;
......@@ -89,7 +90,7 @@ static int g722_decode_frame(AVCodecContext *avctx, void *data,
G722Context *c = avctx->priv_data;
int16_t *out_buf;
int j, ret;
const int skip = 8 - avctx->bits_per_coded_sample;
const int skip = 8 - c->bits_per_codeword;
const int16_t *quantizer_table = low_inv_quants[skip];
GetBitContext gb;
......@@ -149,4 +150,5 @@ AVCodec ff_adpcm_g722_decoder = {
.decode = g722_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.priv_class = &g722_decoder_class,
};
libavcodec/g722enc.c
View file @ b404ab9e
......@@ -139,7 +139,7 @@ static int g722_encode_trellis(AVCodecContext *avctx,
nodes[i][0]->state = c->band[i];
}
for (i = 0; i < buf_size >> 1; i++) {
for (i = 0; i < buf_size; i++) {
int xlow, xhigh;
struct TrellisNode *next[2];
int heap_pos[2] = {0, 0};
......@@ -285,7 +285,7 @@ static int g722_encode_frame(AVCodecContext *avctx,
if (avctx->trellis)
return g722_encode_trellis(avctx, dst, buf_size, data);
for (i = 0; i < buf_size >> 1; i++) {
for (i = 0; i < buf_size; i++) {
int xlow, xhigh, ihigh, ilow;
filter_samples(c, &samples[2*i], &xlow, &xhigh);
ihigh = encode_high(&c->band[1], xhigh);
......
libavcodec/ivi_common.h
View file @ b404ab9e
......@@ -51,7 +51,7 @@ typedef struct {
/// or "7" for custom one
VLC *tab; /// pointer to the table associated with tab_sel
//! the following are used only when tab_sel == 7
/// the following are used only when tab_sel == 7
IVIHuffDesc cust_desc; /// custom Huffman codebook descriptor
VLC cust_tab; /// vlc table for custom codebook
} IVIHuffTab;
......
libavcodec/lsp.c
View file @ b404ab9e
......@@ -75,7 +75,7 @@ void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order)
/**
* @brief decodes polynomial coefficients from LSP
* @param f [out] decoded polynomial coefficients (-0x20000000 <= (3.22) <= 0x1fffffff)
* @param[out] f decoded polynomial coefficients (-0x20000000 <= (3.22) <= 0x1fffffff)
* @param lsp LSP coefficients (-0x8000 <= (0.15) <= 0x7fff)
*/
static void lsp2poly(int* f, const int16_t* lsp, int lp_half_order)
......
libavcodec/mlpdec.c
View file @ b404ab9e
......@@ -45,35 +45,35 @@ static const char* sample_message =
"a sample of this file.";
typedef struct SubStream {
//! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
/// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
uint8_t restart_seen;
//@{
/** restart header data */
//! The type of noise to be used in the rematrix stage.
/// The type of noise to be used in the rematrix stage.
uint16_t noise_type;
//! The index of the first channel coded in this substream.
/// The index of the first channel coded in this substream.
uint8_t min_channel;
//! The index of the last channel coded in this substream.
/// The index of the last channel coded in this substream.
uint8_t max_channel;
//! The number of channels input into the rematrix stage.
/// The number of channels input into the rematrix stage.
uint8_t max_matrix_channel;
//! For each channel output by the matrix, the output channel to map it to
/// For each channel output by the matrix, the output channel to map it to
uint8_t ch_assign[MAX_CHANNELS];
//! Channel coding parameters for channels in the substream
/// Channel coding parameters for channels in the substream
ChannelParams channel_params[MAX_CHANNELS];
//! The left shift applied to random noise in 0x31ea substreams.
/// The left shift applied to random noise in 0x31ea substreams.
uint8_t noise_shift;
//! The current seed value for the pseudorandom noise generator(s).
/// The current seed value for the pseudorandom noise generator(s).
uint32_t noisegen_seed;
//! Set if the substream contains extra info to check the size of VLC blocks.
/// Set if the substream contains extra info to check the size of VLC blocks.
uint8_t data_check_present;
//! Bitmask of which parameter sets are conveyed in a decoding parameter block.
/// Bitmask of which parameter sets are conveyed in a decoding parameter block.
uint8_t param_presence_flags;
#define PARAM_BLOCKSIZE (1 << 7)
#define PARAM_MATRIX (1 << 6)
......@@ -88,32 +88,32 @@ typedef struct SubStream {
//@{
/** matrix data */
//! Number of matrices to be applied.
/// Number of matrices to be applied.
uint8_t num_primitive_matrices;
//! matrix output channel
/// matrix output channel
uint8_t matrix_out_ch[MAX_MATRICES];
//! Whether the LSBs of the matrix output are encoded in the bitstream.
/// Whether the LSBs of the matrix output are encoded in the bitstream.
uint8_t lsb_bypass[MAX_MATRICES];
//! Matrix coefficients, stored as 2.14 fixed point.
/// Matrix coefficients, stored as 2.14 fixed point.
int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
//! Left shift to apply to noise values in 0x31eb substreams.
/// Left shift to apply to noise values in 0x31eb substreams.
uint8_t matrix_noise_shift[MAX_MATRICES];
//@}
//! Left shift to apply to Huffman-decoded residuals.
/// Left shift to apply to Huffman-decoded residuals.
uint8_t quant_step_size[MAX_CHANNELS];
//! number of PCM samples in current audio block
/// number of PCM samples in current audio block
uint16_t blocksize;
//! Number of PCM samples decoded so far in this frame.
/// Number of PCM samples decoded so far in this frame.
uint16_t blockpos;
//! Left shift to apply to decoded PCM values to get final 24-bit output.
/// Left shift to apply to decoded PCM values to get final 24-bit output.
int8_t output_shift[MAX_CHANNELS];
//! Running XOR of all output samples.
/// Running XOR of all output samples.
int32_t lossless_check_data;
} SubStream;
......@@ -122,24 +122,24 @@ typedef struct MLPDecodeContext {
AVCodecContext *avctx;
AVFrame frame;
//! Current access unit being read has a major sync.
/// Current access unit being read has a major sync.
int is_major_sync_unit;
//! Set if a valid major sync block has been read. Otherwise no decoding is possible.
/// Set if a valid major sync block has been read. Otherwise no decoding is possible.
uint8_t params_valid;
//! Number of substreams contained within this stream.
/// Number of substreams contained within this stream.
uint8_t num_substreams;
//! Index of the last substream to decode - further substreams are skipped.
/// Index of the last substream to decode - further substreams are skipped.
uint8_t max_decoded_substream;
//! Stream needs channel reordering to comply with FFmpeg's channel order
/// Stream needs channel reordering to comply with FFmpeg's channel order
uint8_t needs_reordering;
//! number of PCM samples contained in each frame
/// number of PCM samples contained in each frame
int access_unit_size;
//! next power of two above the number of samples in each frame
/// next power of two above the number of samples in each frame
int access_unit_size_pow2;
SubStream substream[MAX_SUBSTREAMS];
......
libavcodec/qcelpdata.h
View file @ b404ab9e
......@@ -40,16 +40,16 @@
typedef struct {
/// @name QCELP excitation codebook parameters
/// @{
uint8_t cbsign[16]; ///!< sign of the codebook gain for each codebook subframe
uint8_t cbgain[16]; ///!< unsigned codebook gain for each codebook subframe
uint8_t cindex[16]; ///!< codebook index for each codebook subframe
uint8_t cbsign[16]; ///< sign of the codebook gain for each codebook subframe
uint8_t cbgain[16]; ///< unsigned codebook gain for each codebook subframe
uint8_t cindex[16]; ///< codebook index for each codebook subframe
/// @}
/// @name QCELP pitch prediction parameters
/// @{
uint8_t plag[4]; ///!< pitch lag for each pitch subframe
uint8_t pfrac[4]; ///!< fractional pitch lag for each pitch subframe
uint8_t pgain[4]; ///!< pitch gain for each pitch subframe
uint8_t plag[4]; ///< pitch lag for each pitch subframe
uint8_t pfrac[4]; ///< fractional pitch lag for each pitch subframe
uint8_t pgain[4]; ///< pitch gain for each pitch subframe
/// @}
/**
......@@ -266,7 +266,7 @@ static const QCELPBitmap qcelp_rate_octave_bitmap[] = {
* the QCELPContext
*/
static const QCELPBitmap * const qcelp_unpacking_bitmaps_per_rate[5] = {
NULL, ///!< for SILENCE rate
NULL, ///< for SILENCE rate
qcelp_rate_octave_bitmap,
qcelp_rate_quarter_bitmap,
qcelp_rate_half_bitmap,
......@@ -274,7 +274,7 @@ static const QCELPBitmap * const qcelp_unpacking_bitmaps_per_rate[5] = {
};
static const uint16_t qcelp_unpacking_bitmaps_lengths[5] = {
0, ///!< for SILENCE rate
0, ///< for SILENCE rate
FF_ARRAY_ELEMS(qcelp_rate_octave_bitmap),
FF_ARRAY_ELEMS(qcelp_rate_quarter_bitmap),
FF_ARRAY_ELEMS(qcelp_rate_half_bitmap),
......
libavcodec/rtjpeg.c
View file @ b404ab9e
......@@ -27,7 +27,7 @@
i = scan[coeff--]; \
block[i] = (c) * quant[i];
//! aligns the bitstream to the give power of two
/// aligns the bitstream to the given power of two
#define ALIGN(a) \
n = (-get_bits_count(gb)) & (a - 1); \
if (n) {skip_bits(gb, n);}
......
libavcodec/utils.c
View file @ b404ab9e
......@@ -1468,8 +1468,8 @@ int av_get_bits_per_sample(enum CodecID codec_id){
case CODEC_ID_ADPCM_SWF:
case CODEC_ID_ADPCM_MS:
case CODEC_ID_ADPCM_YAMAHA:
return 4;
case CODEC_ID_ADPCM_G722:
return 4;
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_S8:
......
libavcodec/version.h
View file @ b404ab9e
......@@ -21,7 +21,7 @@
#define AVCODEC_VERSION_H
#define LIBAVCODEC_VERSION_MAJOR 53
#define LIBAVCODEC_VERSION_MINOR 40
#define LIBAVCODEC_VERSION_MINOR 41
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
......
libavdevice/pulse.c
View file @ b404ab9e
......@@ -23,7 +23,6 @@
* @file
* PulseAudio input using the simple API.
* @author Luca Barbato <lu_zero@gentoo.org>
*
*/
#include <pulse/simple.h>
......
libavdevice/x11grab.c
View file @ b404ab9e
......@@ -31,8 +31,9 @@
/**
* @file
* X11 frame device demuxer by Clemens Fruhwirth <clemens@endorphin.org>
* and Edouard Gomez <ed.gomez@free.fr>.
* X11 frame device demuxer
* @author Clemens Fruhwirth <clemens@endorphin.org>
* @author Edouard Gomez <ed.gomez@free.fr>
*/
#include "config.h"
......
libavformat/avformat.h
View file @ b404ab9e
......@@ -293,7 +293,7 @@ typedef struct AVFormatParameters {
#endif
} AVFormatParameters;
//! Demuxer will use avio_open, no opened file should be provided by the caller.
/// Demuxer will use avio_open, no opened file should be provided by the caller.
#define AVFMT_NOFILE 0x0001
#define AVFMT_NEEDNUMBER 0x0002 /**< Needs '%d' in filename. */
#define AVFMT_SHOW_IDS 0x0008 /**< Show format stream IDs numbers. */
......
libavformat/matroskadec.c
View file @ b404ab9e
......@@ -22,10 +22,10 @@
/**
* @file
* Matroska file demuxer
* by Ronald Bultje <rbultje@ronald.bitfreak.net>
* with a little help from Moritz Bunkus <moritz@bunkus.org>
* totally reworked by Aurelien Jacobs <aurel@gnuage.org>
* Specs available on the Matroska project page: http://www.matroska.org/.
* @author Ronald Bultje <rbultje@ronald.bitfreak.net>
* @author with a little help from Moritz Bunkus <moritz@bunkus.org>
* @author totally reworked by Aurelien Jacobs <aurel@gnuage.org>
* @see specs available on the Matroska project page: http://www.matroska.org/
*/
#include <stdio.h>
......
libavformat/mov.c
View file @ b404ab9e
......@@ -957,6 +957,8 @@ static int mov_read_stco(MOVContext *c, AVIOContext *pb, MOVAtom atom)
entries = avio_rb32(pb);
if (!entries)
return 0;
if (entries >= UINT_MAX/sizeof(int64_t))
return -1;
......@@ -1398,6 +1400,8 @@ static int mov_read_stsc(MOVContext *c, AVIOContext *pb, MOVAtom atom)
av_dlog(c->fc, "track[%i].stsc.entries = %i\n", c->fc->nb_streams-1, entries);
if (!entries)
return 0;
if (entries >= UINT_MAX / sizeof(*sc->stsc_data))
return -1;
sc->stsc_data = av_malloc(entries * sizeof(*sc->stsc_data));
......@@ -1513,6 +1517,8 @@ static int mov_read_stsz(MOVContext *c, AVIOContext *pb, MOVAtom atom)
return -1;
}
if (!entries)
return 0;
if (entries >= UINT_MAX / sizeof(int) || entries >= (UINT_MAX - 4) / field_size)
return -1;
sc->sample_sizes = av_malloc(entries * sizeof(int));
......@@ -1615,6 +1621,8 @@ static int mov_read_ctts(MOVContext *c, AVIOContext *pb, MOVAtom atom)
av_dlog(c->fc, "track[%i].ctts.entries = %i\n", c->fc->nb_streams-1, entries);
if (!entries)
return 0;
if (entries >= UINT_MAX / sizeof(*sc->ctts_data))
return -1;
sc->ctts_data = av_malloc(entries * sizeof(*sc->ctts_data));
......@@ -1675,6 +1683,8 @@ static void mov_build_index(MOVContext *mov, AVStream *st)
current_dts -= sc->dts_shift;
if (!sc->sample_count)
return;
if (sc->sample_count >= UINT_MAX / sizeof(*st->index_entries))
return;
st->index_entries = av_malloc(sc->sample_count*sizeof(*st->index_entries));
......
libavformat/nuv.c
View file @ b404ab9e
......@@ -47,7 +47,7 @@ static int nuv_probe(AVProbeData *p) {
return 0;
}
//! little macro to sanitize packet size
/// little macro to sanitize packet size
#define PKTSIZE(s) (s & 0xffffff)
/**
......
libavformat/oggdec.c
View file @ b404ab9e
......@@ -2,10 +2,9 @@
* Ogg bitstream support
* Luca Barbato <lu_zero@gentoo.org>
* Based on tcvp implementation
*
*/
/**
/*
Copyright (C) 2005 Michael Ahlberg, Måns Rullgård
Permission is hereby granted, free of charge, to any person
......@@ -27,7 +26,7 @@
WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
**/
*/
#include <stdio.h>
#include "oggdec.h"
......
libavformat/utils.c
View file @ b404ab9e
......@@ -2209,10 +2209,10 @@ static int has_decode_delay_been_guessed(AVStream *st)
static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **options)
{
int16_t *samples;
AVCodec *codec;
int got_picture, data_size, ret=0;
int got_picture, ret = 0;
AVFrame picture;
AVPacket pkt = *avpkt;
if(!st->codec->codec){
codec = avcodec_find_decoder(st->codec->codec_id);
......@@ -2223,28 +2223,29 @@ static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **option
return ret;
}
if(!has_codec_parameters(st->codec) || !has_decode_delay_been_guessed(st) ||
(!st->codec_info_nb_frames && st->codec->codec->capabilities & CODEC_CAP_CHANNEL_CONF)) {
while (pkt.size > 0 && ret >= 0 &&
(!has_codec_parameters(st->codec) ||
!has_decode_delay_been_guessed(st) ||
(!st->codec_info_nb_frames && st->codec->codec->capabilities & CODEC_CAP_CHANNEL_CONF))) {
got_picture = 0;
avcodec_get_frame_defaults(&picture);
switch(st->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
avcodec_get_frame_defaults(&picture);
ret = avcodec_decode_video2(st->codec, &picture,
&got_picture, avpkt);
if (got_picture)
st->info->nb_decoded_frames++;
&got_picture, &pkt);
break;
case AVMEDIA_TYPE_AUDIO:
data_size = FFMAX(avpkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE);
samples = av_malloc(data_size);
if (!samples)
goto fail;
ret = avcodec_decode_audio3(st->codec, samples,
&data_size, avpkt);
av_free(samples);
ret = avcodec_decode_audio4(st->codec, &picture, &got_picture, &pkt);
break;
default:
break;
}
if (ret >= 0) {
if (got_picture)
st->info->nb_decoded_frames++;
pkt.data += ret;
pkt.size -= ret;
}
}
fail:
return ret;
......
libavutil/lzo.c
View file @ b404ab9e
......@@ -21,14 +21,14 @@
#include "avutil.h"
#include "common.h"
//! Avoid e.g. MPlayers fast_memcpy, it slows things down here.
/// Avoid e.g. MPlayers fast_memcpy, it slows things down here.
#undef memcpy
#include <string.h>
#include "lzo.h"
//! Define if we may write up to 12 bytes beyond the output buffer.
/// Define if we may write up to 12 bytes beyond the output buffer.
#define OUTBUF_PADDED 1
//! Define if we may read up to 8 bytes beyond the input buffer.
/// Define if we may read up to 8 bytes beyond the input buffer.
#define INBUF_PADDED 1
typedef struct LZOContext {
const uint8_t *in, *in_end;
......
libavutil/lzo.h
View file @ b404ab9e
......@@ -33,13 +33,13 @@
/** @name Error flags returned by av_lzo1x_decode
* \{ */
//! end of the input buffer reached before decoding finished
/// end of the input buffer reached before decoding finished
#define AV_LZO_INPUT_DEPLETED 1
//! decoded data did not fit into output buffer
/// decoded data did not fit into output buffer
#define AV_LZO_OUTPUT_FULL 2
//! a reference to previously decoded data was wrong
/// a reference to previously decoded data was wrong
#define AV_LZO_INVALID_BACKPTR 4
//! a non-specific error in the compressed bitstream
/// a non-specific error in the compressed bitstream
#define AV_LZO_ERROR 8
/** \} */
......
libpostproc/postprocess.c
View file @ b404ab9e
......@@ -644,7 +644,7 @@ static inline void postProcess(const uint8_t src[], int srcStride, uint8_t dst[]
#endif
postProcess_C(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
#endif
#else //CONFIG_RUNTIME_CPUDETECT
#else /* CONFIG_RUNTIME_CPUDETECT */
#if HAVE_MMX2
postProcess_MMX2(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
#elif HAVE_AMD3DNOW
......@@ -656,7 +656,7 @@ static inline void postProcess(const uint8_t src[], int srcStride, uint8_t dst[]
#else
postProcess_C(src, srcStride, dst, dstStride, width, height, QPs, QPStride, isColor, c);
#endif
#endif //!CONFIG_RUNTIME_CPUDETECT
#endif /* !CONFIG_RUNTIME_CPUDETECT */
}
//static void postProcess(uint8_t src[], int srcStride, uint8_t dst[], int dstStride, int width, int height,
......
tests/codec-regression.sh
View file @ b404ab9e
......@@ -352,6 +352,11 @@ do_audio_encoding g723_1.tco "-b:a 6.3k -ac 1 -ar 8000 -acodec g723_1"
do_audio_decoding
fi
if [ -n "$do_g722" ] ; then
do_audio_encoding g722.wav "-b 64k -ac 1 -ar 16000 -acodec g722"
do_audio_decoding
fi
if [ -n "$do_g726" ] ; then
do_audio_encoding g726.wav "-b:a 32k -ac 1 -ar 8000 -acodec g726"
do_audio_decoding
......
tests/fate/h264.mak
View file @ b404ab9e
This diff is collapsed.
tests/ref/acodec/g722 0 → 100644
View file @ b404ab9e
156f63e3391b95020ae882dbae6eccf3 *./tests/data/acodec/g722.wav
47991 ./tests/data/acodec/g722.wav
8f65de513acc08b37a488d6a802b4f00 *./tests/data/g722.acodec.out.wav
stddev: 8860.50 PSNR: 17.38 MAXDIFF:33814 bytes: 191732/ 1058400
tools/patcheck
View file @ b404ab9e
......@@ -55,6 +55,7 @@ hiegrep 'INIT_VLC_USE_STATIC' 'forbidden ancient vlc type' $*
hiegrep '=[-+\*\&] ' 'looks like compound assignment' $*
hiegrep2 '/\*\* *[a-zA-Z0-9].*' '\*/' 'Inconsistently formatted doxygen comment' $*
hiegrep '; */\*\*[^<]' 'Misformatted doxygen comment' $*
hiegrep '//!|/\*!' 'inconsistent doxygen syntax' $*
hiegrep2 '(int|unsigned|static|void)[a-zA-Z0-9 _]*(init|end)[a-zA-Z0-9 _]*\(.*[^;]$' '(av_cold|:\+[^a-zA-Z_])' 'These functions may need av_cold, please review the whole patch for similar functions needing av_cold' $*
......
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