C++ opus_strerror函数代码示例

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本文整理汇总了C++中opus_strerror函数的典型用法代码示例。如果您正苦于以下问题：C++ opus_strerror函数的具体用法？C++ opus_strerror怎么用？C++ opus_strerror使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。

在下文中一共展示了opus_strerror函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: init_audio_encoder

static int init_audio_encoder(CSSession *cs)
{
    int rc = OPUS_OK;
    cs->audio_encoder = opus_encoder_create(cs->audio_encoder_sample_rate,
                                            cs->audio_encoder_channels, OPUS_APPLICATION_AUDIO, &rc);

    if ( rc != OPUS_OK ) {
        LOGGER_ERROR("Error while starting audio encoder: %s", opus_strerror(rc));
        return -1;
    }

    rc = opus_encoder_ctl(cs->audio_encoder, OPUS_SET_BITRATE(cs->audio_encoder_bitrate));

    if ( rc != OPUS_OK ) {
        LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(rc));
        return -1;
    }

    rc = opus_encoder_ctl(cs->audio_encoder, OPUS_SET_COMPLEXITY(10));

    if ( rc != OPUS_OK ) {
        LOGGER_ERROR("Error while setting encoder ctl: %s", opus_strerror(rc));
        return -1;
    }

    return 0;
}

开发者ID:ittner，项目名称:toxcore，代码行数:27，代码来源:codec.c

示例2: toxav_recv_audio

/**
 * @brief Receive decoded audio frame.
 *
 * @param av Handler.
 * @param frame_size The size of dest in frames/samples (one frame/sample is 16 bits or 2 bytes
 *                   and corresponds to one sample of audio.)
 * @param dest Destination of the raw audio (16 bit signed pcm with AUDIO_CHANNELS channels).
 *             Make sure it has enough space for frame_size frames/samples.
 * @return int
 * @retval >=0 Size of received data in frames/samples.
 * @retval ToxAvError On error.
 */
inline__ int toxav_recv_audio ( ToxAv *av, int32_t call_index, int frame_size, int16_t *dest )
{
    if ( !dest ) return ErrorInternal;

    if (cii(call_index, av->msi_session) || !av->calls[call_index].call_active) {
        LOGGER_WARNING("Action on inactive call: %d", call_index);
        return ErrorNoCall;
    }


    CallSpecific *call = &av->calls[call_index];

    uint8_t packet [RTP_PAYLOAD_SIZE];

    int recved_size = toxav_recv_rtp_payload(av, call_index, TypeAudio, packet);

    if ( recved_size == ErrorAudioPacketLost ) {
        int dec_size = opus_decode(call->cs->audio_decoder, NULL, 0, dest, frame_size, 1);

        if ( dec_size < 0 ) {
            LOGGER_WARNING("Decoding error: %s", opus_strerror(dec_size));
            return ErrorInternal;
        } else return dec_size;

    } else if ( recved_size ) {
        int dec_size = opus_decode(call->cs->audio_decoder, packet, recved_size, dest, frame_size, 0);

        if ( dec_size < 0 ) {
            LOGGER_WARNING("Decoding error: %s", opus_strerror(dec_size));
            return ErrorInternal;
        } else return dec_size;
    } else {
        return 0; /* Nothing received */
    }
}

开发者ID:buptfeifei，项目名称:toxcore，代码行数:47，代码来源:toxav.c

示例3: apply_max_bitrate

static void apply_max_bitrate(OpusEncData *d) {
	ms_message("Setting opus codec bitrate to [%i] from network bitrate [%i] with ptime [%i]", d->bitrate, d->max_network_bitrate, d->ptime);
	/* give the bitrate to the encoder if exists*/
	if (d->state) {
		opus_int32 maxBandwidth=0;

		/*tell the target bitrate, opus will choose internally the bandwidth to use*/
		int error = opus_encoder_ctl(d->state, OPUS_SET_BITRATE(d->bitrate));
		if (error != OPUS_OK) {
			ms_error("could not set bit rate to opus encoder: %s", opus_strerror(error));
		}

		/* implement maxplaybackrate parameter, which is constraint on top of bitrate */
		if (d->maxplaybackrate <= 8000) {
			maxBandwidth = OPUS_BANDWIDTH_NARROWBAND;
		} else if (d->maxplaybackrate <= 12000) {
			maxBandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
		} else if (d->maxplaybackrate <= 16000) {
			maxBandwidth = OPUS_BANDWIDTH_WIDEBAND;
		} else if (d->maxplaybackrate <= 24000) {
			maxBandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
		} else {
			maxBandwidth = OPUS_BANDWIDTH_FULLBAND;
		}

		if (maxBandwidth!=0){
			error = opus_encoder_ctl(d->state, OPUS_SET_MAX_BANDWIDTH(maxBandwidth));
			if (error != OPUS_OK) {
				ms_error("could not set max bandwidth to opus encoder: %s", opus_strerror(error));
			}
		}
	}

}

开发者ID:xiaolds，项目名称:VideoCallVoIP，代码行数:34，代码来源:msopus.c

示例4: opus_encoder_create

OpusEncoder *create_audio_encoder(Logger *log, int32_t bit_rate, int32_t sampling_rate, int32_t channel_count)
{
    int status = OPUS_OK;
    OpusEncoder *rc = opus_encoder_create(sampling_rate, channel_count, OPUS_APPLICATION_VOIP, &status);

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while starting audio encoder: %s", opus_strerror(status));
        return NULL;
    }

    status = opus_encoder_ctl(rc, OPUS_SET_BITRATE(bit_rate));

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
        goto FAILURE;
    }

    /* Enable in-band forward error correction in codec */
    status = opus_encoder_ctl(rc, OPUS_SET_INBAND_FEC(1));

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
        goto FAILURE;
    }

    /* Make codec resistant to up to 10% packet loss
     * NOTE This could also be adjusted on the fly, rather than hard-coded,
     *      with feedback from the receiving client.
     */
    status = opus_encoder_ctl(rc, OPUS_SET_PACKET_LOSS_PERC(10));

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
        goto FAILURE;
    }

    /* Set algorithm to the highest complexity, maximizing compression */
    status = opus_encoder_ctl(rc, OPUS_SET_COMPLEXITY(10));

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
        goto FAILURE;
    }

    return rc;

FAILURE:
    opus_encoder_destroy(rc);
    return NULL;
}

开发者ID:GrayHatter，项目名称:toxcore，代码行数:50，代码来源:audio.c

示例5: apply_max_bitrate

static void apply_max_bitrate(OpusEncData *d) {
	ms_message("Setting opus codec bitrate to [%i] from network bitrate [%i] with ptime [%i]", d->bitrate, d->max_network_bitrate, d->ptime);
	/* give the bitrate to the encoder if exists*/
	if (d->state) {
		opus_int32 maxBandwidth;

		int error = opus_encoder_ctl(d->state, OPUS_SET_BITRATE(d->bitrate));
		if (error != OPUS_OK) {
			ms_error("could not set bit rate to opus encoder: %s", opus_strerror(error));
		}

		/* set output sampling rate according to bitrate and RFC section 3.1.1 */
		if (d->bitrate<12000) {
			maxBandwidth = OPUS_BANDWIDTH_NARROWBAND;
		} else if (d->bitrate<20000) {
			maxBandwidth = OPUS_BANDWIDTH_WIDEBAND;
		} else if (d->bitrate<40000) {
			maxBandwidth = OPUS_BANDWIDTH_FULLBAND;
		} else if (d->bitrate<64000) {
			maxBandwidth = OPUS_BANDWIDTH_FULLBAND;
		} else {
			maxBandwidth = OPUS_BANDWIDTH_FULLBAND;
		}

		/* check if selected maxBandwidth is compatible with the maxplaybackrate parameter */
		if (d->maxplaybackrate < 12000) {
			maxBandwidth = OPUS_BANDWIDTH_NARROWBAND;
		} else if (d->maxplaybackrate < 16000) {
			if (maxBandwidth != OPUS_BANDWIDTH_NARROWBAND) {
				maxBandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
			}
		} else if (d->maxplaybackrate < 24000) {
			if (maxBandwidth != OPUS_BANDWIDTH_NARROWBAND) {
				maxBandwidth = OPUS_BANDWIDTH_WIDEBAND;
			}
		} else if (d->maxplaybackrate < 48000) {
			if (maxBandwidth == OPUS_BANDWIDTH_FULLBAND) {
				maxBandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
			}
		}

		error = opus_encoder_ctl(d->state, OPUS_SET_MAX_BANDWIDTH(maxBandwidth));
		if (error != OPUS_OK) {
			ms_error("could not set max bandwidth to opus encoder: %s", opus_strerror(error));
		}
	}

}

开发者ID:biddyweb，项目名称:azfone-ios，代码行数:48，代码来源:msopus.c

示例6: libopus_decode

static int libopus_decode(AVCodecContext *avc, void *frame,
                          int *got_frame_ptr, AVPacket *pkt)
{
    struct libopus_context *opus = avc->priv_data;
    int ret, nb_samples;

    opus->frame.nb_samples = MAX_FRAME_SIZE;
    ret = avc->get_buffer(avc, &opus->frame);
    if (ret < 0) {
        av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }

    if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
        nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
                                             (opus_int16 *)opus->frame.data[0],
                                             opus->frame.nb_samples, 0);
    else
        nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
                                                   (float *)opus->frame.data[0],
                                                   opus->frame.nb_samples, 0);

    if (nb_samples < 0) {
        av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
               opus_strerror(nb_samples));
        return ff_opus_error_to_averror(nb_samples);
    }

    opus->frame.nb_samples = nb_samples;
    *(AVFrame *)frame = opus->frame;
    *got_frame_ptr = 1;
    return pkt->size;
}

开发者ID:JSinglan，项目名称:libav，代码行数:33，代码来源:libopusdec.c

示例7: reconfigure_audio_encoder

bool reconfigure_audio_encoder(Logger *log, OpusEncoder **e, int32_t new_br, int32_t new_sr, uint8_t new_ch,
                               int32_t *old_br, int32_t *old_sr, int32_t *old_ch)
{
    /* Values are checked in toxav.c */
    if (*old_sr != new_sr || *old_ch != new_ch) {
        OpusEncoder *new_encoder = create_audio_encoder(log, new_br, new_sr, new_ch);

        if (new_encoder == NULL) {
            return false;
        }

        opus_encoder_destroy(*e);
        *e = new_encoder;
    } else if (*old_br == new_br) {
        return true; /* Nothing changed */
    }

    int status = opus_encoder_ctl(*e, OPUS_SET_BITRATE(new_br));

    if (status != OPUS_OK) {
        LOGGER_ERROR(log, "Error while setting encoder ctl: %s", opus_strerror(status));
        return false;
    }

    *old_br = new_br;
    *old_sr = new_sr;
    *old_ch = new_ch;

    LOGGER_DEBUG(log, "Reconfigured audio encoder br: %d sr: %d cc:%d", new_br, new_sr, new_ch);
    return true;
}

开发者ID:GrayHatter，项目名称:toxcore，代码行数:31，代码来源:audio.c

示例8: opustolin_framein

static int opustolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
	struct opus_coder_pvt *opvt = pvt->pvt;
	int samples = 0;

	/* Decode */
	ast_debug(3, "[Decoder #%d (%d)] %d samples, %d bytes\n",
		opvt->id,
		opvt->sampling_rate,
		f->samples,
		f->datalen);

	if ((samples = opus_decode(opvt->opus, f->data.ptr, f->datalen, pvt->outbuf.i16, BUFFER_SAMPLES, opvt->fec)) < 0) {
		ast_log(LOG_ERROR, "Error decoding the Opus frame: %s\n", opus_strerror(samples));
		return -1;
	}

	pvt->samples += samples;
	pvt->datalen += samples * 2;

	ast_debug(3, "[Decoder #%d (%d)]   >> Got %d samples, %d bytes\n",
		opvt->id,
		opvt->sampling_rate,
		pvt->samples,
		pvt->datalen);

	return 0;
}

开发者ID:macntouch，项目名称:asterisk-opus-1，代码行数:28，代码来源:codec_opus.c

示例9: toxav_prepare_audio_frame

/**
 * @brief Encode audio frame
 *
 * @param av Handler
 * @param dest dest
 * @param dest_max Max dest size
 * @param frame The frame
 * @param frame_size The frame size
 * @return int
 * @retval ToxAvError On error.
 * @retval >0 On success
 */
int toxav_prepare_audio_frame ( ToxAv *av, int32_t call_index, uint8_t *dest, int dest_max, const int16_t *frame,
                                int frame_size)
{
    if (cii(call_index, av->msi_session) || !av->calls[call_index].call_active) {
        LOGGER_WARNING("Action on inactive call: %d", call_index);
        return ErrorNoCall;
    }

    CallSpecific *call = &av->calls[call_index];
    pthread_mutex_lock(&call->mutex);


    if (!call->call_active) {
        pthread_mutex_unlock(&call->mutex);
        LOGGER_WARNING("Action on inactive call: %d", call_index);
        return ErrorNoCall;
    }

    int32_t rc = opus_encode(call->cs->audio_encoder, frame, frame_size, dest, dest_max);
    pthread_mutex_unlock(&call->mutex);

    if (rc < 0) {
        LOGGER_ERROR("Failed to encode payload: %s\n", opus_strerror(rc));
        return ErrorInternal;
    }

    return rc;
}

开发者ID:13693100472，项目名称:toxcore，代码行数:40，代码来源:toxav.c

示例10: krad_opus_decoder_write

int krad_opus_decoder_write (krad_opus_t *krad_opus,
                             uint8_t *buffer,
                             int length) {

  int i;
  int frames_decoded;

  frames_decoded = 0;

  krad_opus->opus_decoder_error = opus_multistream_decode_float (krad_opus->decoder,
                                  buffer,
                                  length,
                                  krad_opus->interleaved_samples,
                                  2880 * 2,
                                  0);

  if (krad_opus->opus_decoder_error < 0) {
    printke ("Krad Opus decoder error: %s\n",
              opus_strerror (krad_opus->opus_decoder_error));
  } else {
    frames_decoded = krad_opus->opus_decoder_error;
  }

  for (i = 0; i < frames_decoded; i++) {
    krad_opus->samples[0][i] = krad_opus->interleaved_samples[i * 2 + 0];
    krad_opus->samples[1][i] = krad_opus->interleaved_samples[i * 2 + 1];
  }

  krad_ringbuffer_write (krad_opus->ringbuf[0], (char *)krad_opus->samples[0], (frames_decoded * 4) );
  krad_ringbuffer_write (krad_opus->ringbuf[1], (char *)krad_opus->samples[1], (frames_decoded * 4) );

  return 0;
}

开发者ID:kripton，项目名称:krad_radio-1，代码行数:33，代码来源:krad_opus.c

示例11: opus_decoder_construct

static int opus_decoder_construct(struct ast_trans_pvt *pvt, int sampling_rate)
{
	struct opus_coder_pvt *opvt = pvt->pvt;
	int error = 0;

	if (!valid_sampling_rate(sampling_rate)) {
		return -1;
	}

	opvt->sampling_rate = sampling_rate;
	opvt->multiplier = 48000/sampling_rate;
	opvt->fec = USE_FEC;	/* FIXME: should be triggered by chan_sip */

	opvt->opus = opus_decoder_create(sampling_rate, 1, &error);

	if (error != OPUS_OK) {
		ast_log(LOG_ERROR, "Error creating the Opus decoder: %s\n", opus_strerror(error));
		return -1;
	}

	opvt->id = ast_atomic_fetchadd_int(&usage.decoder_id, 1) + 1;

	ast_atomic_fetchadd_int(&usage.decoders, +1);

	ast_debug(3, "Created decoder #%d (opus -> %d)\n", opvt->id, sampling_rate);

	return 0;
}

开发者ID:macntouch，项目名称:asterisk-opus-1，代码行数:28，代码来源:codec_opus.c

示例12: ms_opus_enc_preprocess

static void ms_opus_enc_preprocess(MSFilter *f) {
	int error;

	OpusEncData *d = (OpusEncData *)f->data;
	/* create the encoder */
	d->state = opus_encoder_create(d->samplerate, d->channels, d->application, &error);
	if (error != OPUS_OK) {
		ms_error("Opus encoder creation failed: %s", opus_strerror(error));
		return;
	}

	/* set complexity to 0 for single processor arm devices */
#if defined(__arm__) || defined(_M_ARM)
	if (ms_factory_get_cpu_count(f->factory)==1){
		opus_encoder_ctl(d->state, OPUS_SET_COMPLEXITY(0));
	}else{
		opus_encoder_ctl(d->state, OPUS_SET_COMPLEXITY(5));
	}
#endif
	error = opus_encoder_ctl(d->state, OPUS_SET_PACKET_LOSS_PERC(10));
	if (error != OPUS_OK) {
		ms_error("Could not set default loss percentage to opus encoder: %s", opus_strerror(error));
	}

	/* set the encoder parameters: VBR, IN_BAND_FEC, DTX and bitrate settings */
	ms_opus_enc_set_vbr(f);
	ms_opus_enc_set_inbandfec(f);
	ms_opus_enc_set_dtx(f);
	/* if decoder prefers mono signal, force encoder to output mono signal */
	if (d->stereo == 0) {
		error = opus_encoder_ctl(d->state, OPUS_SET_FORCE_CHANNELS(1));
		if (error != OPUS_OK) {
			ms_error("could not force mono channel to opus encoder: %s", opus_strerror(error));
		}
		if (d->channels == 2) ms_message("Opus encoder configured to encode mono despite it is feed with stereo.");
	}else if (d->channels == 2){
		ms_message("Opus encoder configured to encode stereo.");
	}

	ms_filter_lock(f);
	// set bitrate wasn't call, compute it with the default network bitrate (36000)
	if (d->bitrate==-1) {
		compute_max_bitrate(d, 0);
	}
	apply_max_bitrate(d);
	ms_filter_unlock(f);
}

开发者ID:william30101，项目名称:linphonemedia，代码行数:47，代码来源:msopus.c

示例13: ms_opus_dec_preprocess

static void ms_opus_dec_preprocess(MSFilter *f) {
	int error;
	OpusDecData *d = (OpusDecData *)f->data;
	d->state = opus_decoder_create(d->samplerate, d->channels, &error);
	if (error != OPUS_OK) {
		ms_error("Opus decoder creation failed: %s", opus_strerror(error));
	}
	/* initialise the concealer context */
	d->concealer = ms_concealer_context_new(UINT32_MAX);
}

开发者ID:xiaolds，项目名称:VideoCallVoIP，代码行数:10，代码来源:msopus.c

示例14: opus_encoder_create

OpusEncoder *tc_opus_create_encoder(int sample_rate, int channels, int bitrate)
{
   int err;

   OpusEncoder *encoder = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_AUDIO, &err);
   if (err<0){
      pa_log_error("failed to create an encoder: %s", opus_strerror(err));
      return NULL;
   }

   err = opus_encoder_ctl(encoder, OPUS_SET_BITRATE(bitrate));
   if (err<0)
   {
      pa_log_error("failed to set bitrate: %s", opus_strerror(err));
      return NULL;
   }

   return encoder;
}

开发者ID:indrwthaachen，项目名称:pulseaudio，代码行数:19，代码来源:transcode.c

示例15: ms_opus_enc_preprocess

static void ms_opus_enc_preprocess(MSFilter *f) {
	int error;

	OpusEncData *d = (OpusEncData *)f->data;
	/* create the encoder */
	d->state = opus_encoder_create(d->samplerate, d->channels, d->application, &error);
	if (error != OPUS_OK) {
		ms_error("Opus encoder creation failed: %s", opus_strerror(error));
		return;
	}

	/* set complexity to 0 for arm devices */
#ifdef __arm__
	opus_encoder_ctl(d->state, OPUS_SET_COMPLEXITY(0));
#endif
	error = opus_encoder_ctl(d->state, OPUS_SET_PACKET_LOSS_PERC(10));
	if (error != OPUS_OK) {
		ms_error("Could not set default loss percentage to opus encoder: %s", opus_strerror(error));
	}

	/* set the encoder parameters: VBR, IN_BAND_FEC, DTX and bitrate settings */
	ms_opus_enc_set_vbr(f);
	ms_opus_enc_set_inbandfec(f);
	ms_opus_enc_set_dtx(f);
	/* if decoder prefers mono signal, force encoder to output mono signal */
	if (d->stereo == 0) {
		error = opus_encoder_ctl(d->state, OPUS_SET_FORCE_CHANNELS(1));
		if (error != OPUS_OK) {
			ms_error("could not force mono channel to opus encoder: %s", opus_strerror(error));
		}
	}

	ms_filter_lock(f);
	if (d->ptime==-1) { // set ptime wasn't call, set default:20ms
		d->ptime = 20;
	}
	if (d->bitrate==-1) { // set bitrate wasn't call, compute it with the default network bitrate (36000)
		compute_max_bitrate(d, 0);
	}
	apply_max_bitrate(d);
	ms_filter_unlock(f);
}

开发者ID:biddyweb，项目名称:azfone-ios，代码行数:42，代码来源:msopus.c

示例16: memset

AM_UINT CAudioCodecOpus::encode(AM_U8 *input, AM_UINT inDataSize,
                                AM_U8 *output, AM_UINT *outDataSize)
{
  *outDataSize = 0;
  if (AM_LIKELY(0 == (inDataSize % mEncFrameBytes))) {
    bool isOk = true;
    AM_U8 *out = mEncodeBuf;

    memset(out, 0, sizeof(mEncodeBuf));
    mRepacketizer = opus_repacketizer_init(mRepacketizer);
    for (AM_UINT i = 0; i < (inDataSize / mEncFrameBytes); ++ i) {
      const opus_int16* pcm = (opus_int16*)(input + i * mEncFrameBytes);
      int ret = opus_encode(mEncoder, pcm, mEncFrameSize, out, 4096);
      if (AM_LIKELY(ret > 0)) {
        int retval = opus_repacketizer_cat(mRepacketizer, out, ret);
        if (AM_UNLIKELY(retval != OPUS_OK)) {
          ERROR("Opus repacketizer error: %s", opus_strerror(retval));
          isOk = false;
          break;
        }
        out += ret;
      } else {
        ERROR("Opus encode error: %s", opus_strerror(ret));
        isOk = false;
        break;
      }
    }
    if (AM_LIKELY(isOk)) {
      int ret = opus_repacketizer_out(mRepacketizer, output, 4096);
      if (AM_LIKELY(ret > 0)) {
        *outDataSize = ret;
      } else {
        ERROR("Opus repacketizer error: %s", opus_strerror(ret));
      }
    }
  } else {
    ERROR("Invalid input data length: %u, must be n times of %u",
          inDataSize, mEncFrameBytes);
  }
  return *outDataSize;
}

开发者ID:ShawnOfMisfit，项目名称:ambarella，代码行数:41，代码来源:audio_codec_opus.cpp

示例17: toxav_prepare_audio_frame

/**
 * @brief Encode audio frame
 *
 * @param av Handler
 * @param dest dest
 * @param dest_max Max dest size
 * @param frame The frame
 * @param frame_size The frame size
 * @return int
 * @retval ToxAvError On error.
 * @retval >0 On success
 */
inline__ int toxav_prepare_audio_frame ( ToxAv *av, int32_t call_index, uint8_t *dest, int dest_max,
        const int16_t *frame, int frame_size)
{
    int32_t rc = opus_encode(av->calls[call_index].cs->audio_encoder, frame, frame_size, dest, dest_max);

    if (rc < 0) {
        fprintf(stderr, "Failed to encode payload: %s\n", opus_strerror(rc));
        return ErrorInternal;
    }

    return rc;
}

开发者ID:9cat，项目名称:ProjectTox-Core，代码行数:24，代码来源:toxav.c

示例18: init_audio_decoder

static int init_audio_decoder(CSSession *cs)
{
    int rc;
    cs->audio_decoder = opus_decoder_create(cs->audio_decoder_sample_rate, cs->audio_decoder_channels, &rc );

    if ( rc != OPUS_OK ) {
        LOGGER_ERROR("Error while starting audio decoder: %s", opus_strerror(rc));
        return -1;
    }

    return 0;
}

开发者ID:ittner，项目名称:toxcore，代码行数:12，代码来源:codec.c

示例19: opus_decoder_create

OpusDecoder *tc_opus_create_decoder(int sample_rate, int channels)
{
   int err;

   OpusDecoder * decoder = opus_decoder_create(sample_rate, channels, &err);
   if (err<0) {
      pa_log_error("failed to create decoder: %s", opus_strerror(err));
      return NULL;
   }

   return decoder;
}

开发者ID:indrwthaachen，项目名称:pulseaudio，代码行数:12，代码来源:transcode.c

示例20: init_audio_decoder

int init_audio_decoder(CodecState *cs, uint32_t audio_channels)
{
    int rc;
    cs->audio_decoder = opus_decoder_create(cs->audio_sample_rate, audio_channels, &rc );

    if ( rc != OPUS_OK ) {
        LOGGER_ERROR("Error while starting audio decoder: %s", opus_strerror(rc));
        return -1;
    }

    return 0;
}

开发者ID:kenygia，项目名称:toxcore，代码行数:12，代码来源:media.c

注：本文中的opus_strerror函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。

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C++ opus_strerror函数代码示例

示例1: init_audio_encoder

示例2: toxav_recv_audio

示例3: apply_max_bitrate

示例4: opus_encoder_create

示例5: apply_max_bitrate

示例6: libopus_decode

示例7: reconfigure_audio_encoder

示例8: opustolin_framein

示例9: toxav_prepare_audio_frame

示例10: krad_opus_decoder_write

示例11: opus_decoder_construct

示例12: ms_opus_enc_preprocess

示例13: ms_opus_dec_preprocess

示例14: opus_encoder_create

示例15: ms_opus_enc_preprocess

示例16: memset

示例17: toxav_prepare_audio_frame

示例18: init_audio_decoder

示例19: opus_decoder_create

示例20: init_audio_decoder

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