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cook: use planar sample format
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9af4eaa8ea
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@ -119,9 +119,10 @@ 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, float *out);
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void (*saturate_output)(struct cook *q, float *out);
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AVCodecContext* avctx;
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DSPContext dsp;
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AVFrame frame;
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GetBitContext gb;
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/* stream data */
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@ -876,18 +877,15 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
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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 saturate_output_float(COOKContext *q, int chan, float *out)
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static void saturate_output_float(COOKContext *q, 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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for (j = 0; j < q->samples_per_channel; 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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q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
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-1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
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}
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/**
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* Final part of subpacket decoding:
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* Apply modulated lapped transform, gain compensation,
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@ -898,15 +896,14 @@ static void saturate_output_float(COOKContext *q, int chan, float *out)
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* @param gains_ptr array of current/prev gain pointers
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* @param previous_buffer pointer to the previous buffer to be used for overlapping
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* @param out pointer to the output buffer
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* @param chan 0: left or single channel, 1: right channel
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*/
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static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
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cook_gains *gains_ptr, float *previous_buffer,
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float *out, int chan)
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float *out)
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{
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imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
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if (out)
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q->saturate_output(q, chan, out);
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q->saturate_output(q, out);
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}
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@ -919,7 +916,7 @@ static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
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* @param outbuffer pointer to the outbuffer
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*/
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static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
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const uint8_t *inbuffer, float *outbuffer)
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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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int res;
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@ -942,15 +939,18 @@ static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
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}
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mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
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p->mono_previous_buffer1, outbuffer, p->ch_idx);
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p->mono_previous_buffer1,
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outbuffer ? outbuffer[p->ch_idx] : NULL);
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if (p->num_channels == 2)
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if (p->joint_stereo)
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mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
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p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
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p->mono_previous_buffer2,
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outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
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else
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mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
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p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
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p->mono_previous_buffer2,
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outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
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return 0;
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}
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@ -967,7 +967,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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COOKContext *q = avctx->priv_data;
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float *samples = NULL;
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float **samples = NULL;
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int i, ret;
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int offset = 0;
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int chidx = 0;
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@ -982,7 +982,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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samples = (float *) q->frame.data[0];
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samples = (float **)q->frame.extended_data;
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}
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/* estimate subpacket sizes */
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@ -1099,6 +1099,8 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
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/* Initialize RNG. */
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av_lfg_init(&q->random_state, 0);
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ff_dsputil_init(&q->dsp, avctx);
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while (edata_ptr < edata_ptr_end) {
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/* 8 for mono, 16 for stereo, ? for multichannel
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Swap to right endianness so we don't need to care later on. */
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@ -1274,7 +1276,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
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return AVERROR_PATCHWELCOME;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
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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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@ -1299,4 +1301,6 @@ AVCodec ff_cook_decoder = {
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.decode = cook_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE },
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};
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