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generic-pc: add audio sample conversion

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ffmpeg audiodecoders after version 1.1 deliver audio samples in planar
formats by default instead of packed as before. Additionally, the AC3
decoder delivers now in planar float.
Use libswresample for sample format conversion, this will work with old
(where it hopefully does nothing) and new ffmpeg versions.
Later on, also sample rate and channel layout conversion could be
implemented if desired.
Tested with ffmpeg versions 1.0.6 and 1.2.1.


Origin commit data
------------------
Branch: master
Commit: f4f72f34b9
Author: Stefan Seyfried <seife@tuxbox-git.slipkontur.de>
Date: 2013-06-01 (Sat, 01 Jun 2013)



------------------
This commit was generated by Migit
This commit is contained in:
Stefan Seyfried
2013-06-01 13:43:51 +02:00
parent a5aeedcb84
commit 3d6df50158
2 changed files with 51 additions and 6 deletions
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1
configure.ac
View File

@@ -29,6 +29,7 @@ if test x$BOXTYPE = xgeneric; then
# don't know which version is exactly needed here...
PKG_CHECK_MODULES([AVUTIL], [libavutil])
PKG_CHECK_MODULES([SWSCALE], [libswscale])
PKG_CHECK_MODULES([SWRESAMPLE], [libswresample])
fi
AC_OUTPUT([
Makefile

56
generic-pc/audio.cpp
View File

@@ -16,7 +16,7 @@
*
* cAudio implementation with decoder.
* uses libao <http://www.xiph.org/ao/> for output
* ffmpeg <http://ffmpeg.org> for demuxing / decoding
* ffmpeg <http://ffmpeg.org> for demuxing / decoding / format conversion
*/
#include <cstdio>
@@ -33,6 +33,9 @@
extern "C" {
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
#include <ao/ao.h>
}
/* ffmpeg buf 2k */
@@ -279,6 +282,13 @@ void cAudio::run()
ao_info *ai;
// ao_device *adevice;
// ao_sample_format sformat;
/* resample */
SwrContext *swr = NULL;
uint8_t *obuf = NULL;
int obuf_sz = 0; /* in samples */
int obuf_sz_max = 0;
int o_ch, o_sr; /* output channels and sample rate */
uint64_t o_layout; /* output channels layout */
curr_pts = 0;
av_init_packet(&avpkt);
@@ -324,13 +334,17 @@ void cAudio::run()
lt_info("%s: avcodec_alloc_frame failed\n", __func__);
goto out2;
}
if (sformat.channels != c->channels || sformat.rate != c->sample_rate ||
/* output sample rate, channels, layout could be set here if necessary */
o_ch = c->channels; /* 2 */
o_sr = c->sample_rate; /* 48000 */
o_layout = c->channel_layout; /* AV_CH_LAYOUT_STEREO */
if (sformat.channels != o_ch || sformat.rate != o_sr ||
sformat.byte_format != AO_FMT_NATIVE || sformat.bits != 16 || adevice == NULL)
{
driver = ao_default_driver_id();
sformat.bits = 16;
sformat.channels = c->channels;
sformat.rate = c->sample_rate;
sformat.channels = o_ch;
sformat.rate = o_sr;
sformat.byte_format = AO_FMT_NATIVE;
sformat.matrix = 0;
if (adevice)
@@ -338,7 +352,7 @@ void cAudio::run()
adevice = ao_open_live(driver, &sformat, NULL);
ai = ao_driver_info(driver);
lt_info("%s: changed params ch %d srate %d bits %d adevice %p\n",
__func__, c->channels, c->sample_rate, 16, adevice);;
__func__, o_ch, o_sr, 16, adevice);;
lt_info("libao driver: %d name '%s' short '%s' author '%s'\n",
driver, ai->name, ai->short_name, ai->author);
}
@@ -350,19 +364,49 @@ void cAudio::run()
#endif
lt_info("codec params: sample_fmt %d sample_rate %d channels %d\n",
c->sample_fmt, c->sample_rate, c->channels);
swr = swr_alloc_set_opts(swr,
o_layout, AV_SAMPLE_FMT_S16, o_sr, /* output */
c->channel_layout, c->sample_fmt, c->sample_rate, /* input */
0, NULL);
if (! swr) {
lt_info("could not alloc resample context\n");
goto out3;
}
swr_init(swr);
while (thread_started) {
int gotframe = 0;
if (av_read_frame(avfc, &avpkt) < 0)
break;
avcodec_decode_audio4(c, frame, &gotframe, &avpkt);
if (gotframe && thread_started) {
int out_linesize;
obuf_sz = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) +
frame->nb_samples, o_sr, c->sample_rate, AV_ROUND_UP);
if (obuf_sz > obuf_sz_max) {
lt_info("obuf_sz: %d old: %d\n", obuf_sz, obuf_sz_max);
av_free(obuf);
if (av_samples_alloc(&obuf, &out_linesize, o_ch,
frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0) {
lt_info("av_samples_alloc failed\n");
av_free_packet(&avpkt);
break; /* while (thread_started) */
}
obuf_sz_max = obuf_sz;
}
obuf_sz = swr_convert(swr, &obuf, obuf_sz,
(const uint8_t **)frame->extended_data, frame->nb_samples);
curr_pts = av_frame_get_best_effort_timestamp(frame);
lt_debug("%s: pts 0x%" PRIx64 " %3f\n", __func__, curr_pts, curr_pts/90000.0);
ao_play(adevice, (char*)frame->extended_data[0], frame->linesize[0]);
int o_buf_sz = av_samples_get_buffer_size(&out_linesize, o_ch,
obuf_sz, AV_SAMPLE_FMT_S16, 1);
ao_play(adevice, (char *)obuf, o_buf_sz);
}
av_free_packet(&avpkt);
}
// ao_close(adevice); /* can take long :-( */
av_free(obuf);
swr_free(&swr);
out3:
avcodec_free_frame(&frame);
out2:
avcodec_close(c);