コード例 #1
0
ファイル: dither.c プロジェクト: pansk/cscodec
int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
{
    int ret;
    AudioData *flt_data;

    /* output directly to dst if it is planar */
    if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
        c->s16_data = dst;
    else {
        /* make sure s16_data is large enough for the output */
        ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
        if (ret < 0)
            return ret;
    }

    if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
        /* make sure flt_data is large enough for the input */
        ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
        if (ret < 0)
            return ret;
        flt_data = c->flt_data;

        /* convert input samples to fltp and scale to s16 range */
        ret = ff_audio_convert(c->ac_in, flt_data, src);
        if (ret < 0)
            return ret;
    } else {
        flt_data = src;
    }

    /* check alignment and padding constraints */
    if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
        int ptr_align     = FFMIN(flt_data->ptr_align,     c->s16_data->ptr_align);
        int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
        int aligned_len   = FFALIGN(src->nb_samples, c->ddsp.samples_align);

        if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
            c->quantize      = c->ddsp.quantize;
            c->samples_align = c->ddsp.samples_align;
        } else {
            c->quantize      = quantize_c;
            c->samples_align = 1;
        }
    }

    ret = convert_samples(c, (int16_t **)c->s16_data->data,
                          (float * const *)flt_data->data, src->channels,
                          src->nb_samples);
    if (ret < 0)
        return ret;

    c->s16_data->nb_samples = src->nb_samples;

    /* interleave output to dst if needed */
    if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
        ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
        if (ret < 0)
            return ret;
    } else
        c->s16_data = NULL;

    return 0;
}
コード例 #2
0
ファイル: utils.c プロジェクト: xiaoliang2016/libav
int avresample_convert(AVAudioResampleContext *avr, void **output,
                       int out_plane_size, int out_samples, void **input,
                       int in_plane_size, int in_samples)
{
    AudioData input_buffer;
    AudioData output_buffer;
    AudioData *current_buffer;
    int ret;

    /* reset internal buffers */
    if (avr->in_buffer) {
        avr->in_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->in_buffer,
                                   avr->in_buffer->allocated_channels);
    }
    if (avr->resample_out_buffer) {
        avr->resample_out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->resample_out_buffer,
                                   avr->resample_out_buffer->allocated_channels);
    }
    if (avr->out_buffer) {
        avr->out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->out_buffer,
                                   avr->out_buffer->allocated_channels);
    }

    av_dlog(avr, "[start conversion]\n");

    /* initialize output_buffer with output data */
    if (output) {
        ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
                                 avr->out_channels, out_samples,
                                 avr->out_sample_fmt, 0, "output");
        if (ret < 0)
            return ret;
        output_buffer.nb_samples = 0;
    }

    if (input) {
        /* initialize input_buffer with input data */
        ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
                                 avr->in_channels, in_samples,
                                 avr->in_sample_fmt, 1, "input");
        if (ret < 0)
            return ret;
        current_buffer = &input_buffer;

        if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
                !avr->out_convert_needed && output && out_samples >= in_samples) {
            /* in some rare cases we can copy input to output and upmix
               directly in the output buffer */
            av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
            ret = ff_audio_data_copy(&output_buffer, current_buffer);
            if (ret < 0)
                return ret;
            current_buffer = &output_buffer;
        } else if (avr->mixing_needed || avr->in_convert_needed) {
            /* if needed, copy or convert input to in_buffer, and downmix if
               applicable */
            if (avr->in_convert_needed) {
                ret = ff_audio_data_realloc(avr->in_buffer,
                                            current_buffer->nb_samples);
                if (ret < 0)
                    return ret;
                av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
                                       current_buffer->nb_samples);
                if (ret < 0)
                    return ret;
            } else {
                av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
                if (ret < 0)
                    return ret;
            }
            ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
            if (avr->downmix_needed) {
                av_dlog(avr, "[downmix] in_buffer\n");
                ret = ff_audio_mix(avr->am, avr->in_buffer);
                if (ret < 0)
                    return ret;
            }
            current_buffer = avr->in_buffer;
        }
    } else {
        /* flush resampling buffer and/or output FIFO if input is NULL */
        if (!avr->resample_needed)
            return handle_buffered_output(avr, output ? &output_buffer : NULL,
                                          NULL);
        current_buffer = NULL;
    }

    if (avr->resample_needed) {
        AudioData *resample_out;
        int consumed = 0;

        if (!avr->out_convert_needed && output && out_samples > 0)
            resample_out = &output_buffer;
        else
            resample_out = avr->resample_out_buffer;
        av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
                resample_out->name);
        ret = ff_audio_resample(avr->resample, resample_out,
                                current_buffer, &consumed);
        if (ret < 0)
            return ret;

        /* if resampling did not produce any samples, just return 0 */
        if (resample_out->nb_samples == 0) {
            av_dlog(avr, "[end conversion]\n");
            return 0;
        }

        current_buffer = resample_out;
    }

    if (avr->upmix_needed) {
        av_dlog(avr, "[upmix] %s\n", current_buffer->name);
        ret = ff_audio_mix(avr->am, current_buffer);
        if (ret < 0)
            return ret;
    }

    /* if we resampled or upmixed directly to output, return here */
    if (current_buffer == &output_buffer) {
        av_dlog(avr, "[end conversion]\n");
        return current_buffer->nb_samples;
    }

    if (avr->out_convert_needed) {
        if (output && out_samples >= current_buffer->nb_samples) {
            /* convert directly to output */
            av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
            ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
                                   current_buffer->nb_samples);
            if (ret < 0)
                return ret;

            av_dlog(avr, "[end conversion]\n");
            return output_buffer.nb_samples;
        } else {
            ret = ff_audio_data_realloc(avr->out_buffer,
                                        current_buffer->nb_samples);
            if (ret < 0)
                return ret;
            av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
            ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
                                   current_buffer, current_buffer->nb_samples);
            if (ret < 0)
                return ret;
            current_buffer = avr->out_buffer;
        }
    }

    return handle_buffered_output(avr, output ? &output_buffer : NULL,
                                  current_buffer);
}
コード例 #3
0
ファイル: utils.c プロジェクト: venkatarajasekhar/Qt
int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
                                           uint8_t **output, int out_plane_size,
                                           int out_samples, uint8_t **input,
                                           int in_plane_size, int in_samples)
{
    AudioData input_buffer;
    AudioData output_buffer;
    AudioData *current_buffer;
    int ret, direct_output;

    /* reset internal buffers */
    if (avr->in_buffer) {
        avr->in_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->in_buffer,
                                   avr->in_buffer->allocated_channels);
    }
    if (avr->resample_out_buffer) {
        avr->resample_out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->resample_out_buffer,
                                   avr->resample_out_buffer->allocated_channels);
    }
    if (avr->out_buffer) {
        avr->out_buffer->nb_samples = 0;
        ff_audio_data_set_channels(avr->out_buffer,
                                   avr->out_buffer->allocated_channels);
    }

    av_dlog(avr, "[start conversion]\n");

    /* initialize output_buffer with output data */
    direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0;
    if (output) {
        ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
                                 avr->out_channels, out_samples,
                                 avr->out_sample_fmt, 0, "output");
        if (ret < 0)
            return ret;
        output_buffer.nb_samples = 0;
    }

    if (input) {
        /* initialize input_buffer with input data */
        ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
                                 avr->in_channels, in_samples,
                                 avr->in_sample_fmt, 1, "input");
        if (ret < 0)
            return ret;
        current_buffer = &input_buffer;

        if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
            !avr->out_convert_needed && direct_output && out_samples >= in_samples) {
            /* in some rare cases we can copy input to output and upmix
               directly in the output buffer */
            av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
            ret = ff_audio_data_copy(&output_buffer, current_buffer,
                                     avr->remap_point == REMAP_OUT_COPY ?
                                     &avr->ch_map_info : NULL);
            if (ret < 0)
                return ret;
            current_buffer = &output_buffer;
        } else if (avr->remap_point == REMAP_OUT_COPY &&
                   (!direct_output || out_samples < in_samples)) {
            /* if remapping channels during output copy, we may need to
             * use an intermediate buffer in order to remap before adding
             * samples to the output fifo */
            av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
            ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
                                     &avr->ch_map_info);
            if (ret < 0)
                return ret;
            current_buffer = avr->out_buffer;
        } else if (avr->in_copy_needed || avr->in_convert_needed) {
            /* if needed, copy or convert input to in_buffer, and downmix if
               applicable */
            if (avr->in_convert_needed) {
                ret = ff_audio_data_realloc(avr->in_buffer,
                                            current_buffer->nb_samples);
                if (ret < 0)
                    return ret;
                av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_convert(avr->ac_in, avr->in_buffer,
                                       current_buffer);
                if (ret < 0)
                    return ret;
            } else {
                av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
                ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
                                         avr->remap_point == REMAP_IN_COPY ?
                                         &avr->ch_map_info : NULL);
                if (ret < 0)
                    return ret;
            }
            ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
            if (avr->downmix_needed) {
                av_dlog(avr, "[downmix] in_buffer\n");
                ret = ff_audio_mix(avr->am, avr->in_buffer);
                if (ret < 0)
                    return ret;
            }
            current_buffer = avr->in_buffer;
        }
    } else {
        /* flush resampling buffer and/or output FIFO if input is NULL */
        if (!avr->resample_needed)
            return handle_buffered_output(avr, output ? &output_buffer : NULL,
                                          NULL);
        current_buffer = NULL;
    }

    if (avr->resample_needed) {
        AudioData *resample_out;

        if (!avr->out_convert_needed && direct_output && out_samples > 0)
            resample_out = &output_buffer;
        else
            resample_out = avr->resample_out_buffer;
        av_dlog(avr, "[resample] %s to %s\n",
                current_buffer ? current_buffer->name : "null",
                resample_out->name);
        ret = ff_audio_resample(avr->resample, resample_out,
                                current_buffer);
        if (ret < 0)
            return ret;

        /* if resampling did not produce any samples, just return 0 */
        if (resample_out->nb_samples == 0) {
            av_dlog(avr, "[end conversion]\n");
            return 0;
        }

        current_buffer = resample_out;
    }

    if (avr->upmix_needed) {
        av_dlog(avr, "[upmix] %s\n", current_buffer->name);
        ret = ff_audio_mix(avr->am, current_buffer);
        if (ret < 0)
            return ret;
    }

    /* if we resampled or upmixed directly to output, return here */
    if (current_buffer == &output_buffer) {
        av_dlog(avr, "[end conversion]\n");
        return current_buffer->nb_samples;
    }

    if (avr->out_convert_needed) {
        if (direct_output && out_samples >= current_buffer->nb_samples) {
            /* convert directly to output */
            av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
            ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer);
            if (ret < 0)
                return ret;

            av_dlog(avr, "[end conversion]\n");
            return output_buffer.nb_samples;
        } else {
            ret = ff_audio_data_realloc(avr->out_buffer,
                                        current_buffer->nb_samples);
            if (ret < 0)
                return ret;
            av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
            ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
                                   current_buffer);
            if (ret < 0)
                return ret;
            current_buffer = avr->out_buffer;
        }
    }

    return handle_buffered_output(avr, output ? &output_buffer : NULL,
                                  current_buffer);
}