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cook: use planar sample format

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This commit is contained in:
Justin Ruggles 2012-08-27 10:38:36 -04:00
parent 9af4eaa8ea
commit cbf6ee7823
1 changed files with 22 additions and 18 deletions
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40
libavcodec/cook.c
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@ -119,9 +119,10 @@ typedef struct cook {
void (*interpolate)(struct cook *q, float *buffer, void (*interpolate)(struct cook *q, float *buffer,
int gain_index, int gain_index_next); int gain_index, int gain_index_next);
void (*saturate_output)(struct cook *q, int chan, float *out); void (*saturate_output)(struct cook *q, float *out);
AVCodecContext* avctx; AVCodecContext* avctx;
DSPContext dsp;
AVFrame frame; AVFrame frame;
GetBitContext gb; GetBitContext gb;
/* stream data */ /* stream data */
@ -876,18 +877,15 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
* Saturate the output signal and interleave. * Saturate the output signal and interleave.
* *
* @param q pointer to the COOKContext * @param q pointer to the COOKContext
* @param chan channel to saturate
* @param out pointer to the output vector * @param out pointer to the output vector
*/ */
static void saturate_output_float(COOKContext *q, int chan, float *out) static void saturate_output_float(COOKContext *q, float *out)
{ {
int j; q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
float *output = q->mono_mdct_output + q->samples_per_channel; -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
for (j = 0; j < q->samples_per_channel; j++) {
out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
}
} }
/** /**
* Final part of subpacket decoding: * Final part of subpacket decoding:
* Apply modulated lapped transform, gain compensation, * Apply modulated lapped transform, gain compensation,
@ -898,15 +896,14 @@ static void saturate_output_float(COOKContext *q, int chan, float *out)
* @param gains_ptr array of current/prev gain pointers * @param gains_ptr array of current/prev gain pointers
* @param previous_buffer pointer to the previous buffer to be used for overlapping * @param previous_buffer pointer to the previous buffer to be used for overlapping
* @param out pointer to the output buffer * @param out pointer to the output buffer
* @param chan 0: left or single channel, 1: right channel
*/ */
static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer, static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
cook_gains *gains_ptr, float *previous_buffer, cook_gains *gains_ptr, float *previous_buffer,
float *out, int chan) float *out)
{ {
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer); imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
if (out) if (out)
q->saturate_output(q, chan, out); q->saturate_output(q, out);
} }
@ -919,7 +916,7 @@ static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
* @param outbuffer pointer to the outbuffer * @param outbuffer pointer to the outbuffer
*/ */
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
const uint8_t *inbuffer, float *outbuffer) const uint8_t *inbuffer, float **outbuffer)
{ {
int sub_packet_size = p->size; int sub_packet_size = p->size;
int res; int res;
@ -942,15 +939,18 @@ static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
} }
mlt_compensate_output(q, q->decode_buffer_1, &p->gains1, mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
p->mono_previous_buffer1, outbuffer, p->ch_idx); p->mono_previous_buffer1,
outbuffer ? outbuffer[p->ch_idx] : NULL);
if (p->num_channels == 2) if (p->num_channels == 2)
if (p->joint_stereo) if (p->joint_stereo)
mlt_compensate_output(q, q->decode_buffer_2, &p->gains1, mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
p->mono_previous_buffer2, outbuffer, p->ch_idx + 1); p->mono_previous_buffer2,
outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
else else
mlt_compensate_output(q, q->decode_buffer_2, &p->gains2, mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
p->mono_previous_buffer2, outbuffer, p->ch_idx + 1); p->mono_previous_buffer2,
outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
return 0; return 0;
} }
@ -967,7 +967,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data; const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size; int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data; COOKContext *q = avctx->priv_data;
float *samples = NULL; float **samples = NULL;
int i, ret; int i, ret;
int offset = 0; int offset = 0;
int chidx = 0; int chidx = 0;
@ -982,7 +982,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret; return ret;
} }
samples = (float *) q->frame.data[0]; samples = (float **)q->frame.extended_data;
} }
/* estimate subpacket sizes */ /* estimate subpacket sizes */
@ -1099,6 +1099,8 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
/* Initialize RNG. */ /* Initialize RNG. */
av_lfg_init(&q->random_state, 0); av_lfg_init(&q->random_state, 0);
ff_dsputil_init(&q->dsp, avctx);
while (edata_ptr < edata_ptr_end) { while (edata_ptr < edata_ptr_end) {
/* 8 for mono, 16 for stereo, ? for multichannel /* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */ Swap to right endianness so we don't need to care later on. */
@ -1274,7 +1276,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
return AVERROR_PATCHWELCOME; return AVERROR_PATCHWELCOME;
} }
avctx->sample_fmt = AV_SAMPLE_FMT_FLT; avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (channel_mask) if (channel_mask)
avctx->channel_layout = channel_mask; avctx->channel_layout = channel_mask;
else else
@ -1299,4 +1301,6 @@ AVCodec ff_cook_decoder = {
.decode = cook_decode_frame, .decode = cook_decode_frame,
.capabilities = CODEC_CAP_DR1, .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"), .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
}; };