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generic-pc: fix audio (2nd try)

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martii
2013-06-01 10:37:36 +02:00
parent b8a8c07786
commit a89b6f294e
3 changed files with 53 additions and 8 deletions
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1
Makefile.am
View File

@@ -40,6 +40,7 @@ libstb_hal_test_LDADD += \
@AVFORMAT_LIBS@ \
@AVUTIL_LIBS@ \
@AVCODEC_LIBS@ \
@AVRESAMPLE_LIBS@ \
@SWSCALE_LIBS@
endif
if BOXTYPE_SPARK

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([AVRESAMPLE], [libavresample])
fi
AC_OUTPUT([
Makefile

59
generic-pc/audio.cpp
View File

@@ -34,11 +34,22 @@
extern "C" {
#include <libavformat/avformat.h>
#include <ao/ao.h>
#ifdef MARTII
#include <libavresample/avresample.h>
#include <libavutil/opt.h>
#endif
}
#ifdef MARTII
/* ffmpeg buf 8k */
#define INBUF_SIZE 0x2000
/* my own buf 32k */
#define DMX_BUF_SZ 0x8000
#else
/* ffmpeg buf 2k */
#define INBUF_SIZE 0x0800
/* my own buf 16k */
#define DMX_BUF_SZ 0x4000
#endif
cAudio * audioDecoder = NULL;
extern cDemux *audioDemux;
@@ -279,6 +290,10 @@ void cAudio::run()
ao_info *ai;
// ao_device *adevice;
// ao_sample_format sformat;
#ifdef MARTII
AVAudioResampleContext *avr = NULL;
int e;
#endif
curr_pts = 0;
av_init_packet(&avpkt);
@@ -329,11 +344,7 @@ void cAudio::run()
{
driver = ao_default_driver_id();
sformat.bits = 16;
#ifdef MARTII
sformat.channels = av_get_channel_layout_nb_channels(c->channels);
#else
sformat.channels = c->channels;
#endif
sformat.rate = c->sample_rate;
sformat.byte_format = AO_FMT_NATIVE;
sformat.matrix = 0;
@@ -353,12 +364,21 @@ void cAudio::run()
fprintf(stderr, "\n");
#endif
#ifdef MARTII
lt_info("codec params: sample_fmt %d sample_rate %d channels %d\n",
c->sample_fmt, c->sample_rate, av_get_channel_layout_nb_channels(c->channels));
#else
avr = avresample_alloc_context();
av_opt_set_int(avr, "in_channel_layout", c->channel_layout, 0);
av_opt_set_int(avr, "out_channel_layout", c->channel_layout, 0);
av_opt_set_int(avr, "in_sample_rate", c->sample_rate, 0);
av_opt_set_int(avr, "out_sample_rate", c->sample_rate, 0);
av_opt_set_int(avr, "in_sample_fmt", c->sample_fmt, 0);
av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
e = avresample_open(avr);
if (e < 0) {
lt_info("%s: avresample_open failed: %d\n", __func__, -e);
goto out2;
}
#endif
lt_info("codec params: sample_fmt %d sample_rate %d channels %d\n",
c->sample_fmt, c->sample_rate, c->channels);
#endif
while (thread_started) {
int gotframe = 0;
if (av_read_frame(avfc, &avpkt) < 0)
@@ -367,7 +387,24 @@ void cAudio::run()
if (gotframe && thread_started) {
curr_pts = av_frame_get_best_effort_timestamp(frame);
lt_debug("%s: pts 0x%" PRIx64 " %3f\n", __func__, curr_pts, curr_pts/90000.0);
#ifdef MARTII
uint8_t *output = NULL;
int out_linesize;
int out_samples = avresample_available(avr)
+ av_rescale_rnd(avresample_get_delay(avr) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
e = av_samples_alloc(&output, &out_linesize, c->channels, out_samples, AV_SAMPLE_FMT_S16, 1);
if (e < 0) {
lt_info("%s: av_samples_alloc failed: %d\n", __func__, -e);
goto out2;
}
out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
&frame->data[0], frame->linesize[0], frame->nb_samples);
ao_play(adevice, (char *)output, out_samples * 2 /* 16 bit */ * c->channels);
av_freep(&output);
#else
ao_play(adevice, (char*)frame->extended_data[0], frame->linesize[0]);
#endif
}
av_free_packet(&avpkt);
}
@@ -376,6 +413,12 @@ void cAudio::run()
out2:
avcodec_close(c);
out:
#ifdef MARTII
if (avr) {
avresample_close(avr);
avresample_free(&avr);
}
#endif
avformat_close_input(&avfc);
av_free(pIOCtx->buffer);
av_free(pIOCtx);