示例#1
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#2
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;
}
示例#3
0
文件: dither.c 项目: pansk/cscodec
DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
                               enum AVSampleFormat out_fmt,
                               enum AVSampleFormat in_fmt,
                               int channels, int sample_rate)
{
    AVLFG seed_gen;
    DitherContext *c;
    int ch;

    if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
            av_get_bytes_per_sample(in_fmt) <= 2) {
        av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
               av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
        return NULL;
    }

    c = av_mallocz(sizeof(*c));
    if (!c)
        return NULL;

    if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
            sample_rate != 48000 && sample_rate != 44100) {
        av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
               "for triangular_ns dither. using triangular_hp instead.\n");
        avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
    }
    c->method = avr->dither_method;
    dither_init(&c->ddsp, c->method);

    if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
        if (sample_rate == 48000) {
            c->ns_coef_b = ns_48_coef_b;
            c->ns_coef_a = ns_48_coef_a;
        } else {
            c->ns_coef_b = ns_44_coef_b;
            c->ns_coef_a = ns_44_coef_a;
        }
    }

    /* Either s16 or s16p output format is allowed, but s16p is used
       internally, so we need to use a temp buffer and interleave if the output
       format is s16 */
    if (out_fmt != AV_SAMPLE_FMT_S16P) {
        c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
                                          "dither s16 buffer");
        if (!c->s16_data)
            goto fail;

        c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
                                           channels, sample_rate);
        if (!c->ac_out)
            goto fail;
    }

    if (in_fmt != AV_SAMPLE_FMT_FLTP) {
        c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
                                          "dither flt buffer");
        if (!c->flt_data)
            goto fail;

        c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
                                          channels, sample_rate);
        if (!c->ac_in)
            goto fail;
    }

    c->state = av_mallocz(channels * sizeof(*c->state));
    if (!c->state)
        goto fail;
    c->channels = channels;

    /* calculate thresholds for turning off dithering during periods of
       silence to avoid replacing digital silence with quiet dither noise */
    c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
    c->mute_reset_threshold  = c->mute_dither_threshold * 4;

    /* initialize dither states */
    av_lfg_init(&seed_gen, 0xC0FFEE);
    for (ch = 0; ch < channels; ch++) {
        DitherState *state = &c->state[ch];
        state->mute = c->mute_reset_threshold + 1;
        state->seed = av_lfg_get(&seed_gen);
        generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
    }

    return c;

fail:
    ff_dither_free(&c);
    return NULL;
}
示例#4
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;
}