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https://github.com/xenia-project/FFmpeg.git
synced 2024-11-25 12:40:01 +00:00
cook: output float samples instead of converting to int16
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69e7ad8dbc
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c25df22365
@ -124,7 +124,7 @@ typedef struct cook {
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void (* interpolate) (struct cook *q, float* buffer,
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int gain_index, int gain_index_next);
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void (* saturate_output) (struct cook *q, int chan, int16_t *out);
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void (* saturate_output) (struct cook *q, int chan, float *out);
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AVCodecContext* avctx;
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GetBitContext gb;
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@ -229,7 +229,7 @@ static av_cold int init_cook_mlt(COOKContext *q) {
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q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
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/* Initialize the MDCT. */
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if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) {
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if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0/32768.0)) {
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av_free(q->mlt_window);
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return -1;
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}
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@ -867,22 +867,18 @@ decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
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}
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/**
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* Saturate the output signal to signed 16bit integers.
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* Saturate the output signal and interleave.
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*
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* @param q pointer to the COOKContext
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* @param chan channel to saturate
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* @param out pointer to the output vector
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*/
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static void
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saturate_output_float (COOKContext *q, int chan, int16_t *out)
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static void saturate_output_float(COOKContext *q, int chan, float *out)
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{
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int j;
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float *output = q->mono_mdct_output + q->samples_per_channel;
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/* Clip and convert floats to 16 bits.
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*/
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for (j = 0; j < q->samples_per_channel; j++) {
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out[chan + q->nb_channels * j] =
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av_clip_int16(lrintf(output[j]));
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out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
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}
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}
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@ -902,7 +898,7 @@ saturate_output_float (COOKContext *q, int chan, int16_t *out)
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static inline void
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mlt_compensate_output(COOKContext *q, float *decode_buffer,
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cook_gains *gains_ptr, float *previous_buffer,
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int16_t *out, int chan)
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float *out, int chan)
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{
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imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
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q->saturate_output (q, chan, out);
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@ -917,7 +913,9 @@ mlt_compensate_output(COOKContext *q, float *decode_buffer,
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* @param inbuffer pointer to the inbuffer
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* @param outbuffer pointer to the outbuffer
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*/
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static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
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static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
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const uint8_t *inbuffer, float *outbuffer)
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{
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int sub_packet_size = p->size;
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/* packet dump */
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// for (i=0 ; i<sub_packet_size ; i++) {
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@ -991,12 +989,13 @@ static int cook_decode_frame(AVCodecContext *avctx,
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q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
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q->subpacket[i].ch_idx = chidx;
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av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
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decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
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decode_subpacket(q, &q->subpacket[i], buf + offset, data);
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offset += q->subpacket[i].size;
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chidx += q->subpacket[i].num_channels;
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av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
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}
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*data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;
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*data_size = q->nb_channels * q->samples_per_channel *
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av_get_bytes_per_sample(avctx->sample_fmt);
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/* Discard the first two frames: no valid audio. */
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if (avctx->frame_number < 2) *data_size = 0;
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@ -1240,7 +1239,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
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return -1;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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if (channel_mask)
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avctx->channel_layout = channel_mask;
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else
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