mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-25 08:42:39 +00:00
avconv: split audio transcoding out of output_packet().
This commit is contained in:
Anton Khirnov
parent
78162b4ea2
commit
ded28ba35b
209
avconv.c
209
avconv.c
@ -1615,6 +1615,109 @@ static void rate_emu_sleep(InputStream *ist)
|
||||
}
|
||||
}
|
||||
|
||||
static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
|
||||
{
|
||||
static unsigned int samples_size = 0;
|
||||
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;
|
||||
|
||||
ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
|
||||
pkt);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
pkt->data += ret;
|
||||
pkt->size -= 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 0;
|
||||
}
|
||||
|
||||
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);
|
||||
|
||||
// preprocess audio (volume)
|
||||
if (audio_volume != 256) {
|
||||
switch (ist->st->codec->sample_fmt) {
|
||||
case AV_SAMPLE_FMT_U8:
|
||||
{
|
||||
uint8_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
|
||||
*volp++ = av_clip_uint8(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_S16:
|
||||
{
|
||||
int16_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int v = ((*volp) * audio_volume + 128) >> 8;
|
||||
*volp++ = av_clip_int16(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_S32:
|
||||
{
|
||||
int32_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
|
||||
*volp++ = av_clipl_int32(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_FLT:
|
||||
{
|
||||
float *volp = samples;
|
||||
float scale = audio_volume / 256.f;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
*volp++ *= scale;
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_DBL:
|
||||
{
|
||||
double *volp = samples;
|
||||
double scale = audio_volume / 256.;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
*volp++ *= scale;
|
||||
}
|
||||
break;
|
||||
}
|
||||
default:
|
||||
av_log(NULL, AV_LOG_FATAL,
|
||||
"Audio volume adjustment on sample format %s is not supported.\n",
|
||||
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
|
||||
exit_program(1);
|
||||
}
|
||||
}
|
||||
|
||||
rate_emu_sleep(ist);
|
||||
|
||||
for (i = 0; i < nb_output_streams; i++) {
|
||||
OutputStream *ost = &output_streams[i];
|
||||
|
||||
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);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* pkt = NULL means EOF (needed to flush decoder buffers) */
|
||||
static int output_packet(InputStream *ist, int ist_index,
|
||||
OutputStream *ost_table, int nb_ostreams,
|
||||
@ -1625,7 +1728,6 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
int ret = 0, i;
|
||||
int got_output;
|
||||
void *buffer_to_free = NULL;
|
||||
static unsigned int samples_size= 0;
|
||||
AVSubtitle subtitle, *subtitle_to_free;
|
||||
int64_t pkt_pts = AV_NOPTS_VALUE;
|
||||
#if CONFIG_AVFILTER
|
||||
@ -1634,7 +1736,6 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
float quality;
|
||||
|
||||
AVPacket avpkt;
|
||||
int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
|
||||
|
||||
if(ist->next_pts == AV_NOPTS_VALUE)
|
||||
ist->next_pts= ist->pts;
|
||||
@ -1656,8 +1757,6 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
|
||||
//while we have more to decode or while the decoder did output something on EOF
|
||||
while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) {
|
||||
uint8_t *decoded_data_buf;
|
||||
int decoded_data_size;
|
||||
AVFrame *decoded_frame, *filtered_frame;
|
||||
handle_eof:
|
||||
ist->pts= ist->next_pts;
|
||||
@ -1667,38 +1766,19 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
"Multiple frames in a packet from stream %d\n", pkt->stream_index);
|
||||
ist->showed_multi_packet_warning=1;
|
||||
|
||||
/* decode the packet if needed */
|
||||
decoded_frame = filtered_frame = NULL;
|
||||
decoded_data_buf = NULL; /* fail safe */
|
||||
decoded_data_size= 0;
|
||||
subtitle_to_free = NULL;
|
||||
switch(ist->st->codec->codec_type) {
|
||||
case AVMEDIA_TYPE_AUDIO:{
|
||||
if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
|
||||
samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
|
||||
av_free(samples);
|
||||
samples= av_malloc(samples_size);
|
||||
}
|
||||
decoded_data_size= samples_size;
|
||||
/* XXX: could avoid copy if PCM 16 bits with same
|
||||
endianness as CPU */
|
||||
ret = avcodec_decode_audio3(ist->st->codec, samples, &decoded_data_size,
|
||||
&avpkt);
|
||||
// XXX temporary hack, will be turned to a switch() once all codec
|
||||
// types are split out
|
||||
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
|
||||
ret = transcode_audio(ist, &avpkt, &got_output);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
avpkt.data += ret;
|
||||
avpkt.size -= 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 */
|
||||
continue;
|
||||
}
|
||||
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);
|
||||
break;}
|
||||
continue;
|
||||
}
|
||||
|
||||
/* decode the packet if needed */
|
||||
decoded_frame = filtered_frame = NULL;
|
||||
subtitle_to_free = NULL;
|
||||
switch(ist->st->codec->codec_type) {
|
||||
case AVMEDIA_TYPE_VIDEO:
|
||||
if (!(decoded_frame = avcodec_alloc_frame()))
|
||||
return AVERROR(ENOMEM);
|
||||
@ -1743,64 +1823,6 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
return -1;
|
||||
}
|
||||
|
||||
// preprocess audio (volume)
|
||||
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
|
||||
if (audio_volume != 256) {
|
||||
switch (ist->st->codec->sample_fmt) {
|
||||
case AV_SAMPLE_FMT_U8:
|
||||
{
|
||||
uint8_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
|
||||
*volp++ = av_clip_uint8(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_S16:
|
||||
{
|
||||
int16_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int v = ((*volp) * audio_volume + 128) >> 8;
|
||||
*volp++ = av_clip_int16(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_S32:
|
||||
{
|
||||
int32_t *volp = samples;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
|
||||
*volp++ = av_clipl_int32(v);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_FLT:
|
||||
{
|
||||
float *volp = samples;
|
||||
float scale = audio_volume / 256.f;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
*volp++ *= scale;
|
||||
}
|
||||
break;
|
||||
}
|
||||
case AV_SAMPLE_FMT_DBL:
|
||||
{
|
||||
double *volp = samples;
|
||||
double scale = audio_volume / 256.;
|
||||
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
|
||||
*volp++ *= scale;
|
||||
}
|
||||
break;
|
||||
}
|
||||
default:
|
||||
av_log(NULL, AV_LOG_FATAL,
|
||||
"Audio volume adjustment on sample format %s is not supported.\n",
|
||||
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
|
||||
exit_program(1);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* frame rate emulation */
|
||||
rate_emu_sleep(ist);
|
||||
|
||||
@ -1846,9 +1868,6 @@ static int output_packet(InputStream *ist, int ist_index,
|
||||
|
||||
av_assert0(ist->decoding_needed);
|
||||
switch(ost->st->codec->codec_type) {
|
||||
case AVMEDIA_TYPE_AUDIO:
|
||||
do_audio_out(os, ost, ist, decoded_data_buf, decoded_data_size);
|
||||
break;
|
||||
case AVMEDIA_TYPE_VIDEO:
|
||||
#if CONFIG_AVFILTER
|
||||
if (ost->picref->video && !ost->frame_aspect_ratio)
|
||||
|
||||
Loading…
Reference in New Issue
Block a user