static void uninit(struct af_instance *af)
{
struct af_resample *s = af->priv;
if (s->avrctx)
avresample_close(s->avrctx);
avresample_free(&s->avrctx);
if (s->avrctx_out)
avresample_close(s->avrctx_out);
avresample_free(&s->avrctx_out);
}
void aubio_source_avcodec_reset_resampler(aubio_source_avcodec_t * s, uint_t multi) {
if ( (multi != s->multi) || (s->avr == NULL) ) {
int64_t input_layout = av_get_default_channel_layout(s->input_channels);
uint_t output_channels = multi ? s->input_channels : 1;
int64_t output_layout = av_get_default_channel_layout(output_channels);
if (s->avr != NULL) {
avresample_close( s->avr );
av_free ( s->avr );
s->avr = NULL;
}
AVAudioResampleContext *avr = s->avr;
avr = avresample_alloc_context();
av_opt_set_int(avr, "in_channel_layout", input_layout, 0);
av_opt_set_int(avr, "out_channel_layout", output_layout, 0);
av_opt_set_int(avr, "in_sample_rate", s->input_samplerate, 0);
av_opt_set_int(avr, "out_sample_rate", s->samplerate, 0);
av_opt_set_int(avr, "in_sample_fmt", s->avCodecCtx->sample_fmt, 0);
av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
int err;
if ( ( err = avresample_open(avr) ) < 0) {
char errorstr[256];
av_strerror (err, errorstr, sizeof(errorstr));
AUBIO_ERR("source_avcodec: Could not open AVAudioResampleContext for %s (%s)\n",
s->path, errorstr);
//goto beach;
return;
}
s->avr = avr;
s->multi = multi;
}
}
static av_cold void opus_decode_flush(AVCodecContext *ctx)
{
OpusContext *c = ctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
#if CONFIG_SWRESAMPLE
swr_close(s->swr);
#elif CONFIG_AVRESAMPLE
avresample_close(s->avr);
#endif
av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
int update_resampler_configuration( AVAudioResampleContext *avr,
uint64_t out_channel_layout, int out_sample_rate, enum AVSampleFormat out_sample_fmt,
uint64_t in_channel_layout, int in_sample_rate, enum AVSampleFormat in_sample_fmt,
int *input_planes, int *input_block_align )
{
/* Reopen the resampler. */
avresample_close( avr );
av_opt_set_int( avr, "in_channel_layout", in_channel_layout, 0 );
av_opt_set_int( avr, "in_sample_fmt", in_sample_fmt, 0 );
av_opt_set_int( avr, "in_sample_rate", in_sample_rate, 0 );
av_opt_set_int( avr, "out_channel_layout", out_channel_layout, 0 );
av_opt_set_int( avr, "out_sample_fmt", out_sample_fmt, 0 );
av_opt_set_int( avr, "out_sample_rate", out_sample_rate, 0 );
av_opt_set_int( avr, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0 );
if( avresample_open( avr ) < 0 )
return -1;
/* Set up the number of planes and the block alignment of input audio frame. */
int input_channels = av_get_channel_layout_nb_channels( in_channel_layout );
if( av_sample_fmt_is_planar( in_sample_fmt ) )
{
*input_planes = input_channels;
*input_block_align = av_get_bytes_per_sample( in_sample_fmt );
}
else
{
*input_planes = 1;
*input_block_align = av_get_bytes_per_sample( in_sample_fmt ) * input_channels;
}
return 0;
}
void AudioLoader::closeAudioFile() {
if (!_demuxCtx) {
return;
}
#if HAVE_AVRESAMPLE
if (_convertCtxAv) {
avresample_close(_convertCtxAv);
avresample_free(&_convertCtxAv);
}
#elif HAVE_SWRESAMPLE
if (_convertCtx) {
swr_free(&_convertCtx);
}
#endif
// Close the codec
avcodec_close(_audioCtx);
// Close the audio file
avformat_close_input(&_demuxCtx);
// free AVPacket
av_free_packet(&_packet);
_demuxCtx = 0;
_audioCtx = 0;
}
void del_aubio_source_avcodec(aubio_source_avcodec_t * s){
if (!s) return;
if (s->output != NULL) {
av_free(s->output);
}
if (s->avr != NULL) {
avresample_close( s->avr );
av_free ( s->avr );
}
s->avr = NULL;
if (s->avFrame != NULL) {
avcodec_free_frame( &(s->avFrame) );
}
s->avFrame = NULL;
if (s->avCodecCtx != NULL) {
avcodec_close ( s->avCodecCtx );
}
s->avCodecCtx = NULL;
if (s->avFormatCtx != NULL) {
avformat_close_input ( &(s->avFormatCtx) );
}
s->avFrame = NULL;
s->avFormatCtx = NULL;
AUBIO_FREE(s);
}
void avresample_free(AVAudioResampleContext **avr)
{
if (!*avr)
return;
avresample_close(*avr);
av_opt_free(*avr);
av_freep(avr);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
ResampleContext *s = ctx->priv;
char buf1[64], buf2[64];
int ret;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
if (inlink->channel_layout == outlink->channel_layout &&
inlink->sample_rate == outlink->sample_rate &&
(inlink->format == outlink->format ||
(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
av_get_planar_sample_fmt(inlink->format) ==
av_get_planar_sample_fmt(outlink->format))))
return 0;
if (!(s->avr = avresample_alloc_context()))
return AVERROR(ENOMEM);
if (s->options) {
AVDictionaryEntry *e = NULL;
while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
av_opt_set_dict(s->avr, &s->options);
}
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
s->next_in_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ResampleContext *s = ctx->priv;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
}
/*
* Resamples a frame if needed and pushes the audio data along
* with a timestamp onto the framelist.
*/
static void
resampleframe(AVFrame *avframe)
{
AVAudioResampleContext *avr;
struct frame *frame;
uint8_t *output;
size_t outputlen;
int linesize, nr, samples;
if (avframe->format != out_sample_fmt ||
avframe->channel_layout != out_channel_layout ||
avframe->sample_rate != out_sample_rate) {
avr = avresample_alloc_context();
if (avr == NULL) {
errx(1, "(%s:%d) avresample_alloc_context",
__FILE__, __LINE__);
}
av_opt_set_int(avr, "in_channel_layout",
avframe->channel_layout, 0);
av_opt_set_int(avr, "out_channel_layout", out_channel_layout,
0);
av_opt_set_int(avr, "in_sample_rate", avframe->sample_rate, 0);
av_opt_set_int(avr, "out_sample_rate", out_sample_rate, 0);
av_opt_set_int(avr, "in_sample_fmt", avframe->format, 0);
av_opt_set_int(avr, "out_sample_fmt", out_sample_fmt, 0);
nr = avresample_open(avr);
if (nr < 0) {
avresample_free(&avr);
return;
}
outputlen = av_samples_get_buffer_size(&linesize, out_channels,
avframe->nb_samples, out_sample_fmt, 0);
output = xmalloc(outputlen);
samples = avresample_convert(avr, &output, linesize,
avframe->nb_samples, avframe->data, avframe->linesize[0],
avframe->nb_samples);
outputlen = samples * out_channels *
av_get_bytes_per_sample(out_sample_fmt);
avresample_close(avr);
avresample_free(&avr);
} else {
outputlen = av_samples_get_buffer_size(NULL,
avfmt->streams[sti]->codec->channels,
avframe->nb_samples, avframe->format, 1);
output = xmalloc(outputlen);
memcpy(output, avframe->data[0], outputlen);
}
frame = xmalloc(sizeof(struct frame));
frame->data = output;
frame->size = outputlen;
frame->pts = avframe->pkt_pts;
flpush(frame);
}
AudioResamplerFfmpeg::~AudioResamplerFfmpeg() {
if (_context) {
#ifdef HAVE_SWRESAMPLE_H
swr_free(&_context);
#elif HAVE_AVRESAMPLE_H
avresample_close(_context);
avresample_free(&_context);
#else
audio_resample_close(_context);
#endif
}
}
AudioDecoderThread::~AudioDecoderThread()
{
if (m_pResampleContext) {
#ifdef LIBAVRESAMPLE_VERSION
avresample_close(m_pResampleContext);
avresample_free(&m_pResampleContext);
#else
audio_resample_close(m_pResampleContext);
#endif
m_pResampleContext = 0;
}
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
ResampleContext *s = ctx->priv;
char buf1[64], buf2[64];
int ret;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
if (inlink->channel_layout == outlink->channel_layout &&
inlink->sample_rate == outlink->sample_rate &&
inlink->format == outlink->format)
return 0;
if (!(s->avr = avresample_alloc_context()))
return AVERROR(ENOMEM);
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
/* if both the input and output formats are s16 or u8, use s16 as
the internal sample format */
if (av_get_bytes_per_sample(inlink->format) <= 2 &&
av_get_bytes_per_sample(outlink->format) <= 2)
av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
return 0;
}
void FFmpegEncoderClose(struct FFmpegEncoder* encoder) {
if (!encoder->context) {
return;
}
av_write_trailer(encoder->context);
avio_close(encoder->context->pb);
if (encoder->audioCodec) {
av_free(encoder->postaudioBuffer);
if (encoder->audioBuffer) {
av_free(encoder->audioBuffer);
}
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_free(&encoder->audioFrame);
#else
avcodec_free_frame(&encoder->audioFrame);
#endif
avcodec_close(encoder->audio);
if (encoder->resampleContext) {
#ifdef USE_LIBAVRESAMPLE
avresample_close(encoder->resampleContext);
#else
swr_free(&encoder->resampleContext);
#endif
}
if (encoder->absf) {
#ifdef FFMPEG_USE_NEW_BSF
av_bsf_free(&encoder->absf);
#else
av_bitstream_filter_close(encoder->absf);
encoder->absf = 0;
#endif
}
}
#if LIBAVCODEC_VERSION_MAJOR >= 55
av_frame_free(&encoder->videoFrame);
#else
avcodec_free_frame(&encoder->videoFrame);
#endif
avcodec_close(encoder->video);
sws_freeContext(encoder->scaleContext);
encoder->scaleContext = NULL;
avformat_free_context(encoder->context);
encoder->context = 0;
}
void
audio_decoder_destroy(struct audio_decoder *ad)
{
mp_send_cmd(ad->ad_mp, &ad->ad_mp->mp_audio, MB_CTRL_EXIT);
hts_thread_join(&ad->ad_tid);
avcodec_free_frame(&ad->ad_frame);
if(ad->ad_avr != NULL) {
avresample_close(ad->ad_avr);
avresample_free(&ad->ad_avr);
}
free(ad);
}
uint_t aubio_source_avcodec_close(aubio_source_avcodec_t * s) {
if (s->avr != NULL) {
avresample_close( s->avr );
av_free ( s->avr );
}
s->avr = NULL;
if (s->avCodecCtx != NULL) {
avcodec_close ( s->avCodecCtx );
}
s->avCodecCtx = NULL;
if (s->avFormatCtx != NULL) {
avformat_close_input ( &(s->avFormatCtx) );
}
s->avFormatCtx = NULL;
return AUBIO_OK;
}
void
audio_decoder_destroy(struct audio_decoder *ad)
{
mp_send_cmd(ad->ad_mp, &ad->ad_mp->mp_audio, MB_CTRL_EXIT);
hts_thread_join(&ad->ad_tid);
mq_flush(ad->ad_mp, &ad->ad_mp->mp_audio, 1);
av_frame_free(&ad->ad_frame);
if(ad->ad_avr != NULL) {
avresample_close(ad->ad_avr);
avresample_free(&ad->ad_avr);
}
audio_cleanup_spdif_muxer(ad);
free(ad);
}
uint_t aubio_source_avcodec_seek (aubio_source_avcodec_t * s, uint_t pos) {
int64_t resampled_pos = (uint_t)ROUND(pos * (s->input_samplerate * 1. / s->samplerate));
int64_t min_ts = MAX(resampled_pos - 2000, 0);
int64_t max_ts = MIN(resampled_pos + 2000, INT64_MAX);
int seek_flags = AVSEEK_FLAG_FRAME | AVSEEK_FLAG_ANY;
int ret = avformat_seek_file(s->avFormatCtx, s->selected_stream,
min_ts, resampled_pos, max_ts, seek_flags);
if (ret < 0) {
AUBIO_ERR("Failed seeking to %d in file %s", pos, s->path);
}
// reset read status
s->eof = 0;
s->read_index = 0;
s->read_samples = 0;
// reset the AVAudioResampleContext
avresample_close(s->avr);
avresample_open(s->avr);
return ret;
}
static av_cold void opus_decode_flush(AVCodecContext *ctx)
{
OpusContext *c = ctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
avresample_close(s->avr);
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
void CMixer::FlushBuffers()
{
if (m_in_samplerate != m_out_samplerate) {
// Close Resample Context
avresample_close(m_pAVRCxt);
int ret = 0;
// Open Resample Context
ret = avresample_open(m_pAVRCxt);
if (ret < 0) {
TRACE(_T("Mixer: avresample_open failed\n"));
return;
}
// Set Matrix on the context
ret = avresample_set_matrix(m_pAVRCxt, m_matrix_dbl, m_in_layout);
if (ret < 0) {
TRACE(_T("Mixer: avresample_set_matrix failed\n"));
return;
}
}
}
OMX_ERRORTYPE LibavAudioDec::AudioFilterInstanceDeInit()
{
OMX_ERRORTYPE ret = OMX_ErrorNone;
if(codecContext) {
avcodec_close(codecContext);
av_free(codecContext);
codecContext = NULL;
}
if(frame) {
av_frame_free(&frame);
frame = NULL;
}
if(avr) {
avresample_close(avr);
av_free(avr);
avr = NULL;
}
errorCnt = 0;
return ret;
}
void AudioLoader::closeAudioFile() {
if (!_demuxCtx) {
return;
}
if (_convertCtxAv) {
avresample_close(_convertCtxAv);
avresample_free(&_convertCtxAv);
}
// Close the codec
if (!_audioCtx) avcodec_close(_audioCtx);
// Close the audio file
if (!_demuxCtx) avformat_close_input(&_demuxCtx);
// free AVPacket
// TODO: use a variable for whether _packet is initialized or not
av_free_packet(&_packet);
_demuxCtx = 0;
_audioCtx = 0;
_streams.clear();
}
static void
audio_process_audio(audio_decoder_t *ad, media_buf_t *mb)
{
const audio_class_t *ac = ad->ad_ac;
AVFrame *frame = ad->ad_frame;
media_pipe_t *mp = ad->ad_mp;
media_queue_t *mq = &mp->mp_audio;
int r;
int got_frame;
AVPacket avpkt;
int offset = 0;
if(mb->mb_skip || mb->mb_stream != mq->mq_stream)
return;
while(offset < mb->mb_size) {
if(mb->mb_cw == NULL) {
frame->sample_rate = mb->mb_rate;
frame->format = AV_SAMPLE_FMT_S16;
switch(mb->mb_channels) {
case 1:
frame->channel_layout = AV_CH_LAYOUT_MONO;
frame->nb_samples = mb->mb_size / 2;
break;
case 2:
frame->channel_layout = AV_CH_LAYOUT_STEREO;
frame->nb_samples = mb->mb_size / 4;
break;
default:
abort();
}
frame->data[0] = mb->mb_data;
frame->linesize[0] = 0;
r = mb->mb_size;
got_frame = 1;
} else {
av_init_packet(&avpkt);
avpkt.data = mb->mb_data + offset;
avpkt.size = mb->mb_size - offset;
r = avcodec_decode_audio4(mb->mb_cw->codec_ctx, frame,
&got_frame, &avpkt);
if(r < 0)
return;
if(frame->sample_rate == 0)
frame->sample_rate = mb->mb_cw->codec_ctx->sample_rate;
if(frame->sample_rate == 0)
return;
if(mp->mp_stats)
mp_set_mq_meta(mq, mb->mb_cw->codec, mb->mb_cw->codec_ctx);
}
if(offset == 0 && mb->mb_pts != AV_NOPTS_VALUE) {
int od = 0, id = 0;
if(ad->ad_avr != NULL) {
od = avresample_available(ad->ad_avr) *
1000000LL / ad->ad_out_sample_rate;
id = avresample_get_delay(ad->ad_avr) *
1000000LL / frame->sample_rate;
}
ad->ad_pts = mb->mb_pts - od - id;
ad->ad_epoch = mb->mb_epoch;
// printf("od=%-20d id=%-20d PTS=%-20ld oPTS=%-20ld\n",
// od, id, mb->mb_pts, pts);
if(mb->mb_drive_clock)
mp_set_current_time(mp, mb->mb_pts - ad->ad_delay,
mb->mb_epoch, mb->mb_delta);
}
offset += r;
if(got_frame) {
if(frame->sample_rate != ad->ad_in_sample_rate ||
frame->format != ad->ad_in_sample_format ||
frame->channel_layout != ad->ad_in_channel_layout) {
ad->ad_in_sample_rate = frame->sample_rate;
ad->ad_in_sample_format = frame->format;
ad->ad_in_channel_layout = frame->channel_layout;
ac->ac_reconfig(ad);
if(ad->ad_avr == NULL)
ad->ad_avr = avresample_alloc_context();
else
avresample_close(ad->ad_avr);
av_opt_set_int(ad->ad_avr, "in_sample_fmt",
ad->ad_in_sample_format, 0);
av_opt_set_int(ad->ad_avr, "in_sample_rate",
ad->ad_in_sample_rate, 0);
av_opt_set_int(ad->ad_avr, "in_channel_layout",
ad->ad_in_channel_layout, 0);
av_opt_set_int(ad->ad_avr, "out_sample_fmt",
ad->ad_out_sample_format, 0);
av_opt_set_int(ad->ad_avr, "out_sample_rate",
ad->ad_out_sample_rate, 0);
av_opt_set_int(ad->ad_avr, "out_channel_layout",
ad->ad_out_channel_layout, 0);
char buf1[128];
char buf2[128];
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, ad->ad_in_channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, ad->ad_out_channel_layout);
TRACE(TRACE_DEBUG, "Audio",
"Converting from [%s %dHz %s] to [%s %dHz %s]",
buf1, ad->ad_in_sample_rate,
av_get_sample_fmt_name(ad->ad_in_sample_format),
buf2, ad->ad_out_sample_rate,
av_get_sample_fmt_name(ad->ad_out_sample_format));
if(avresample_open(ad->ad_avr)) {
TRACE(TRACE_ERROR, "AudioQueue", "Unable to open resampler");
avresample_free(&ad->ad_avr);
}
}
if(ad->ad_avr != NULL)
avresample_convert(ad->ad_avr, NULL, 0, 0,
frame->data, frame->linesize[0],
frame->nb_samples);
}
}
}
static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
struct mp_audio *out)
{
struct af_resample *s = af->priv;
enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
if (in_samplefmt == AV_SAMPLE_FMT_NONE || out_samplefmt == AV_SAMPLE_FMT_NONE)
return AF_ERROR;
avresample_close(s->avrctx);
avresample_close(s->avrctx_out);
s->ctx.out_rate = out->rate;
s->ctx.in_rate = in->rate;
s->ctx.out_format = out->format;
s->ctx.in_format = in->format;
s->ctx.out_channels= out->channels;
s->ctx.in_channels = in->channels;
s->ctx.filter_size = s->opts.filter_size;
s->ctx.phase_shift = s->opts.phase_shift;
s->ctx.linear = s->opts.linear;
s->ctx.cutoff = s->opts.cutoff;
av_opt_set_int(s->avrctx, "filter_size", s->ctx.filter_size, 0);
av_opt_set_int(s->avrctx, "phase_shift", s->ctx.phase_shift, 0);
av_opt_set_int(s->avrctx, "linear_interp", s->ctx.linear, 0);
av_opt_set_double(s->avrctx, "cutoff", s->ctx.cutoff, 0);
if (parse_avopts(s->avrctx, s->avopts) < 0) {
mp_msg(MSGT_VFILTER, MSGL_FATAL,
"af_lavrresample: could not set opts: '%s'\n", s->avopts);
return AF_ERROR;
}
struct mp_chmap map_in = in->channels;
struct mp_chmap map_out = out->channels;
// Try not to do any remixing if at least one is "unknown".
if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out)) {
mp_chmap_set_unknown(&map_in, map_in.num);
mp_chmap_set_unknown(&map_out, map_out.num);
}
// unchecked: don't take any channel reordering into account
uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);
av_opt_set_int(s->avrctx, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s->avrctx, "in_sample_rate", s->ctx.in_rate, 0);
av_opt_set_int(s->avrctx, "out_sample_rate", s->ctx.out_rate, 0);
av_opt_set_int(s->avrctx, "in_sample_fmt", in_samplefmt, 0);
av_opt_set_int(s->avrctx, "out_sample_fmt", out_samplefmt, 0);
struct mp_chmap in_lavc;
mp_chmap_from_lavc(&in_lavc, in_ch_layout);
mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
struct mp_chmap out_lavc;
mp_chmap_from_lavc(&out_lavc, out_ch_layout);
mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
// Same configuration; we just reorder.
av_opt_set_int(s->avrctx_out, "in_channel_layout", out_ch_layout, 0);
av_opt_set_int(s->avrctx_out, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s->avrctx_out, "in_sample_fmt", out_samplefmt, 0);
av_opt_set_int(s->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
av_opt_set_int(s->avrctx_out, "in_sample_rate", s->ctx.out_rate, 0);
av_opt_set_int(s->avrctx_out, "out_sample_rate", s->ctx.out_rate, 0);
#if USE_SET_CHANNEL_MAPPING
// API has weird requirements, quoting avresample.h:
// * This function can only be called when the allocated context is not open.
// * Also, the input channel layout must have already been set.
avresample_set_channel_mapping(s->avrctx, s->reorder_in);
avresample_set_channel_mapping(s->avrctx_out, s->reorder_out);
#endif
if (avresample_open(s->avrctx) < 0 ||
avresample_open(s->avrctx_out) < 0)
{
mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot open "
"Libavresample Context. \n");
return AF_ERROR;
}
return AF_OK;
}
void swr_free(struct SwrContext **s)
{
avresample_close(*s);
*s = NULL;
}
static void
audio_process_audio(audio_decoder_t *ad, media_buf_t *mb)
{
const audio_class_t *ac = ad->ad_ac;
AVFrame *frame = ad->ad_frame;
media_pipe_t *mp = ad->ad_mp;
media_queue_t *mq = &mp->mp_audio;
int r;
int got_frame;
if(mb->mb_skip || mb->mb_stream != mq->mq_stream)
return;
while(mb->mb_size) {
if(mb->mb_cw == NULL) {
frame->sample_rate = mb->mb_rate;
frame->format = AV_SAMPLE_FMT_S16;
switch(mb->mb_channels) {
case 1:
frame->channel_layout = AV_CH_LAYOUT_MONO;
frame->nb_samples = mb->mb_size / 2;
break;
case 2:
frame->channel_layout = AV_CH_LAYOUT_STEREO;
frame->nb_samples = mb->mb_size / 4;
break;
default:
abort();
}
frame->data[0] = mb->mb_data;
frame->linesize[0] = 0;
r = mb->mb_size;
got_frame = 1;
} else {
media_codec_t *mc = mb->mb_cw;
AVCodecContext *ctx = mc->ctx;
if(mc->codec_id != ad->ad_in_codec_id) {
AVCodec *codec = avcodec_find_decoder(mc->codec_id);
TRACE(TRACE_DEBUG, "audio", "Codec changed to %s (0x%x)",
codec ? codec->name : "???", mc->codec_id);
ad->ad_in_codec_id = mc->codec_id;
ad->ad_in_sample_rate = 0;
audio_cleanup_spdif_muxer(ad);
ad->ad_mode = ac->ac_get_mode != NULL ?
ac->ac_get_mode(ad, mc->codec_id,
ctx ? ctx->extradata : NULL,
ctx ? ctx->extradata_size : 0) : AUDIO_MODE_PCM;
if(ad->ad_mode == AUDIO_MODE_SPDIF) {
audio_setup_spdif_muxer(ad, codec, mq);
} else if(ad->ad_mode == AUDIO_MODE_CODED) {
hts_mutex_lock(&mp->mp_mutex);
ac->ac_deliver_coded_locked(ad, mb->mb_data, mb->mb_size,
mb->mb_pts, mb->mb_epoch);
hts_mutex_unlock(&mp->mp_mutex);
return;
}
}
if(ad->ad_spdif_muxer != NULL) {
mb->mb_pkt.stream_index = 0;
ad->ad_pts = mb->mb_pts;
ad->ad_epoch = mb->mb_epoch;
mb->mb_pts = AV_NOPTS_VALUE;
mb->mb_dts = AV_NOPTS_VALUE;
av_write_frame(ad->ad_spdif_muxer, &mb->mb_pkt);
avio_flush(ad->ad_spdif_muxer->pb);
return;
}
if(ad->ad_mode == AUDIO_MODE_CODED) {
ad->ad_pts = mb->mb_pts;
ad->ad_epoch = mb->mb_epoch;
}
if(ctx == NULL) {
AVCodec *codec = avcodec_find_decoder(mc->codec_id);
assert(codec != NULL); // Checked in libav.c
ctx = mc->ctx = avcodec_alloc_context3(codec);
if(ad->ad_stereo_downmix)
ctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
if(avcodec_open2(mc->ctx, codec, NULL) < 0) {
av_freep(&mc->ctx);
return;
}
}
r = avcodec_decode_audio4(ctx, frame, &got_frame, &mb->mb_pkt);
if(r < 0)
return;
if(frame->sample_rate == 0) {
frame->sample_rate = ctx->sample_rate;
if(frame->sample_rate == 0 && mb->mb_cw->fmt_ctx)
frame->sample_rate = mb->mb_cw->fmt_ctx->sample_rate;
if(frame->sample_rate == 0) {
if(!ad->ad_sample_rate_fail) {
ad->ad_sample_rate_fail = 1;
TRACE(TRACE_ERROR, "Audio",
"Unable to determine sample rate");
}
return;
}
}
if(frame->channel_layout == 0) {
frame->channel_layout = av_get_default_channel_layout(ctx->channels);
if(frame->channel_layout == 0) {
if(!ad->ad_channel_layout_fail) {
ad->ad_channel_layout_fail = 1;
TRACE(TRACE_ERROR, "Audio",
"Unable to map %d channels to channel layout");
}
return;
}
}
if(mp->mp_stats)
mp_set_mq_meta(mq, ctx->codec, ctx);
}
if(mb->mb_pts != PTS_UNSET) {
int od = 0, id = 0;
if(ad->ad_avr != NULL) {
od = avresample_available(ad->ad_avr) *
1000000LL / ad->ad_out_sample_rate;
id = avresample_get_delay(ad->ad_avr) *
1000000LL / frame->sample_rate;
}
ad->ad_pts = mb->mb_pts - od - id;
ad->ad_epoch = mb->mb_epoch;
if(mb->mb_drive_clock)
mp_set_current_time(mp, mb->mb_pts - ad->ad_delay,
mb->mb_epoch, mb->mb_delta);
mb->mb_pts = PTS_UNSET; // No longer valid
}
mb->mb_data += r;
mb->mb_size -= r;
if(got_frame) {
if(frame->sample_rate != ad->ad_in_sample_rate ||
frame->format != ad->ad_in_sample_format ||
frame->channel_layout != ad->ad_in_channel_layout ||
ad->ad_want_reconfig) {
ad->ad_want_reconfig = 0;
ad->ad_in_sample_rate = frame->sample_rate;
ad->ad_in_sample_format = frame->format;
ad->ad_in_channel_layout = frame->channel_layout;
ac->ac_reconfig(ad);
if(ad->ad_avr == NULL)
ad->ad_avr = avresample_alloc_context();
else
avresample_close(ad->ad_avr);
av_opt_set_int(ad->ad_avr, "in_sample_fmt",
ad->ad_in_sample_format, 0);
av_opt_set_int(ad->ad_avr, "in_sample_rate",
ad->ad_in_sample_rate, 0);
av_opt_set_int(ad->ad_avr, "in_channel_layout",
ad->ad_in_channel_layout, 0);
av_opt_set_int(ad->ad_avr, "out_sample_fmt",
ad->ad_out_sample_format, 0);
av_opt_set_int(ad->ad_avr, "out_sample_rate",
ad->ad_out_sample_rate, 0);
av_opt_set_int(ad->ad_avr, "out_channel_layout",
ad->ad_out_channel_layout, 0);
char buf1[128];
char buf2[128];
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, ad->ad_in_channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, ad->ad_out_channel_layout);
TRACE(TRACE_DEBUG, "Audio",
"Converting from [%s %dHz %s] to [%s %dHz %s]",
buf1, ad->ad_in_sample_rate,
av_get_sample_fmt_name(ad->ad_in_sample_format),
buf2, ad->ad_out_sample_rate,
av_get_sample_fmt_name(ad->ad_out_sample_format));
if(avresample_open(ad->ad_avr)) {
TRACE(TRACE_ERROR, "Audio", "Unable to open resampler");
avresample_free(&ad->ad_avr);
}
prop_set(mp->mp_prop_ctrl, "canAdjustVolume", PROP_SET_INT, 1);
if(ac->ac_set_volume != NULL)
ac->ac_set_volume(ad, ad->ad_vol_scale);
}
if(ad->ad_avr != NULL) {
avresample_convert(ad->ad_avr, NULL, 0, 0,
frame->data, frame->linesize[0],
frame->nb_samples);
} else {
int delay = 1000000LL * frame->nb_samples / frame->sample_rate;
usleep(delay);
}
}
}
}
int hb_audio_resample_update(hb_audio_resample_t *resample)
{
if (resample == NULL)
{
hb_error("hb_audio_resample_update: resample is NULL");
return 1;
}
int ret, resample_changed;
resample->resample_needed =
(resample->out.sample_fmt != resample->in.sample_fmt ||
resample->out.channel_layout != resample->in.channel_layout);
resample_changed =
(resample->resample_needed &&
(resample->resample.sample_fmt != resample->in.sample_fmt ||
resample->resample.channel_layout != resample->in.channel_layout ||
resample->resample.center_mix_level != resample->in.center_mix_level ||
resample->resample.surround_mix_level != resample->in.surround_mix_level));
if (resample_changed || (resample->resample_needed &&
resample->avresample == NULL))
{
if (resample->avresample == NULL)
{
resample->avresample = avresample_alloc_context();
if (resample->avresample == NULL)
{
hb_error("hb_audio_resample_update: avresample_alloc_context() failed");
return 1;
}
av_opt_set_int(resample->avresample, "out_sample_fmt",
resample->out.sample_fmt, 0);
av_opt_set_int(resample->avresample, "out_channel_layout",
resample->out.channel_layout, 0);
av_opt_set_int(resample->avresample, "matrix_encoding",
resample->out.matrix_encoding, 0);
av_opt_set_int(resample->avresample, "normalize_mix_level",
resample->out.normalize_mix_level, 0);
}
else if (resample_changed)
{
avresample_close(resample->avresample);
}
av_opt_set_int(resample->avresample, "in_sample_fmt",
resample->in.sample_fmt, 0);
av_opt_set_int(resample->avresample, "in_channel_layout",
resample->in.channel_layout, 0);
av_opt_set_double(resample->avresample, "center_mix_level",
resample->in.center_mix_level, 0);
av_opt_set_double(resample->avresample, "surround_mix_level",
resample->in.surround_mix_level, 0);
if ((ret = avresample_open(resample->avresample)))
{
char err_desc[64];
av_strerror(ret, err_desc, 63);
hb_error("hb_audio_resample_update: avresample_open() failed (%s)",
err_desc);
// avresample won't open, start over
avresample_free(&resample->avresample);
return ret;
}
resample->resample.sample_fmt = resample->in.sample_fmt;
resample->resample.channel_layout = resample->in.channel_layout;
resample->resample.channels =
av_get_channel_layout_nb_channels(resample->in.channel_layout);
resample->resample.center_mix_level = resample->in.center_mix_level;
resample->resample.surround_mix_level = resample->in.surround_mix_level;
}
return 0;
}
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
if (avresample_is_open(avr)) {
av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n");
return AVERROR(EINVAL);
}
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
(!avr->downmix_needed && (avr->mix_matrix ||
avr->in_channel_layout != avr->out_channel_layout));
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
/* select internal sample format if not specified by the user */
if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
(avr->mixing_needed || avr->resample_needed)) {
enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
av_get_bytes_per_sample(out_fmt));
if (max_bps <= 2) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
} else {
if (max_bps <= 4) {
if (in_fmt == AV_SAMPLE_FMT_S32P ||
out_fmt == AV_SAMPLE_FMT_S32P) {
if (in_fmt == AV_SAMPLE_FMT_FLTP ||
out_fmt == AV_SAMPLE_FMT_FLTP) {
/* if one is s32 and the other is flt, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
} else {
/* if one is s32 and the other is s32, s16, or u8, use s32 */
avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
}
} else {
/* if one is flt and the other is flt, s16 or u8, use flt */
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
} else {
/* if either is dbl, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
}
}
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
}
/* treat all mono as planar for easier comparison */
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
/* we may need to add an extra conversion in order to remap channels if
the output format is not planar */
if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
!av_sample_fmt_is_planar(avr->out_sample_fmt)) {
avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
}
/* set sample format conversion parameters */
if (avr->resample_needed || avr->mixing_needed)
avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
else
avr->in_convert_needed = avr->use_channel_map &&
!av_sample_fmt_is_planar(avr->out_sample_fmt);
if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
(avr->use_channel_map && avr->resample_needed));
if (avr->use_channel_map) {
if (avr->in_copy_needed) {
avr->remap_point = REMAP_IN_COPY;
av_dlog(avr, "remap channels during in_copy\n");
} else if (avr->in_convert_needed) {
avr->remap_point = REMAP_IN_CONVERT;
av_dlog(avr, "remap channels during in_convert\n");
} else if (avr->out_convert_needed) {
avr->remap_point = REMAP_OUT_CONVERT;
av_dlog(avr, "remap channels during out_convert\n");
} else {
avr->remap_point = REMAP_OUT_COPY;
av_dlog(avr, "remap channels during out_copy\n");
}
#ifdef DEBUG
{
int ch;
av_dlog(avr, "output map: ");
if (avr->ch_map_info.do_remap)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "copy map: ");
if (avr->ch_map_info.do_copy)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "zero map: ");
if (avr->ch_map_info.do_zero)
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
else
av_dlog(avr, "n/a");
av_dlog(avr, "\n");
av_dlog(avr, "input map: ");
for (ch = 0; ch < avr->in_channels; ch++)
av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
av_dlog(avr, "\n");
}
#endif
} else
avr->remap_point = REMAP_NONE;
/* allocate buffers */
if (avr->in_copy_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
if (!avr->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
1024, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
avr->in_sample_fmt, avr->in_channels,
avr->in_sample_rate,
avr->remap_point == REMAP_IN_CONVERT);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
avr->out_channels,
avr->out_sample_rate,
avr->remap_point == REMAP_OUT_CONVERT);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
avr->am = ff_audio_mix_alloc(avr);
if (!avr->am) {
ret = AVERROR(ENOMEM);
goto error;
}
}
return 0;
error:
avresample_close(avr);
return ret;
}
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
avr->am->matrix ||
(avr->out_channels == avr->in_channels &&
avr->in_channel_layout != avr->out_channel_layout);
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
/* set sample format conversion parameters */
/* override user-requested internal format to avoid unexpected failures
TODO: support more internal formats */
if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
avr->in_sample_fmt != avr->internal_sample_fmt;
if (avr->resample_needed || avr->mixing_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
/* allocate buffers */
if (avr->mixing_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
if (!avr->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
0, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
avr->in_sample_fmt, avr->in_channels);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
avr->out_channels);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
ret = ff_audio_mix_init(avr);
if (ret < 0)
goto error;
}
return 0;
error:
avresample_close(avr);
return ret;
}
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
(!avr->downmix_needed && (avr->am->matrix ||
avr->in_channel_layout != avr->out_channel_layout));
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
/* select internal sample format if not specified by the user */
if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
(avr->mixing_needed || avr->resample_needed)) {
enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
av_get_bytes_per_sample(out_fmt));
if (max_bps <= 2) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
} else {
if (max_bps <= 4) {
if (in_fmt == AV_SAMPLE_FMT_S32P ||
out_fmt == AV_SAMPLE_FMT_S32P) {
if (in_fmt == AV_SAMPLE_FMT_FLTP ||
out_fmt == AV_SAMPLE_FMT_FLTP) {
/* if one is s32 and the other is flt, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
} else {
/* if one is s32 and the other is s32, s16, or u8, use s32 */
avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
}
} else {
/* if one is flt and the other is flt, s16 or u8, use flt */
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
} else {
/* if either is dbl, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
}
}
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
}
/* set sample format conversion parameters */
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
avr->in_sample_fmt != avr->internal_sample_fmt;
if (avr->resample_needed || avr->mixing_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
/* allocate buffers */
if (avr->mixing_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
if (!avr->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
0, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
avr->in_sample_fmt, avr->in_channels);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
avr->out_channels);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
ret = ff_audio_mix_init(avr);
if (ret < 0)
goto error;
}
return 0;
error:
avresample_close(avr);
return ret;
}