示例#1
0
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);
}
示例#2
0
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;
  }
}
示例#3
0
文件: opusdec.c 项目: mpc-hc/FFmpeg
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);
    }
}
示例#4
0
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;
}
示例#5
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;
}
示例#6
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);
}
示例#7
0
void avresample_free(AVAudioResampleContext **avr)
{
    if (!*avr)
        return;
    avresample_close(*avr);
    av_opt_free(*avr);
    av_freep(avr);
}
示例#8
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 ||
        (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;
}
示例#9
0
static av_cold void uninit(AVFilterContext *ctx)
{
    ResampleContext *s = ctx->priv;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }
}
示例#10
0
文件: player.c 项目: jgmp/libplayer
/*
 * 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
    }
}
示例#12
0
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;
    }
}
示例#13
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;
}
示例#14
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;
}
示例#15
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);
}
示例#16
0
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;
}
示例#17
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);
  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);
}
示例#18
0
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;
}
示例#19
0
文件: opusdec.c 项目: dholm/libav
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);
    }
}
示例#20
0
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;
        }
    }
}
示例#21
0
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;
}
示例#22
0
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();
}
示例#23
0
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);
    }
  }
}
示例#24
0
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;
}
示例#25
0
void swr_free(struct SwrContext **s)
{
    avresample_close(*s);
    *s = NULL;
}
示例#26
0
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);
      }
    }
  }
}
示例#27
0
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;
}
示例#28
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;
}
示例#29
0
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;
}
示例#30
0
文件: utils.c 项目: Flameeyes/libav
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;
}