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ffmpeg
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avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()

This commit is contained in:
Justin Ruggles 2011-11-21 17:41:49 -05:00
parent 6d23d19729
commit d1241ff3b2
1 changed files with 26 additions and 29 deletions
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55
avconv.c
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@ -137,8 +137,6 @@ 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 {
@ -541,7 +539,6 @@ 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();
@ -737,14 +734,11 @@ 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;
@ -752,6 +746,9 @@ 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);
@ -1620,39 +1617,40 @@ 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 (!(decoded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
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) {
av_freep(&decoded_frame);
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;
@ -1713,8 +1711,7 @@ 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;
}