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https://github.com/FFmpeg/FFmpeg.git
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avcodec/g723_1: add support for stereo files
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06a436a224
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62dbcb7ddf
@ -116,9 +116,7 @@ typedef struct FCBParam {
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int pulse_sign[PULSE_MAX];
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} FCBParam;
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typedef struct g723_1_context {
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AVClass *class;
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typedef struct G723_1_ChannelContext {
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G723_1_Subframe subframe[4];
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enum FrameType cur_frame_type;
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enum FrameType past_frame_type;
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@ -144,8 +142,6 @@ typedef struct g723_1_context {
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int reflection_coef;
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int pf_gain; ///< formant postfilter
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///< gain scaling unit memory
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int postfilter;
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int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
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/* encoder */
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@ -158,6 +154,13 @@ typedef struct g723_1_context {
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int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
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int16_t harmonic_mem[PITCH_MAX];
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} G723_1_ChannelContext;
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typedef struct G723_1_Context {
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AVClass *class;
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int postfilter;
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G723_1_ChannelContext ch[2];
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} G723_1_Context;
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@ -42,11 +42,15 @@
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static av_cold int g723_1_decode_init(AVCodecContext *avctx)
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{
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G723_1_Context *p = avctx->priv_data;
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G723_1_Context *s = avctx->priv_data;
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G723_1_ChannelContext *p = &s->ch[0];
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avctx->channel_layout = AV_CH_LAYOUT_MONO;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channels = 1;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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if (avctx->channels < 1 || avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
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return AVERROR(EINVAL);
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}
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avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
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p->pf_gain = 1 << 12;
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memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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@ -65,7 +69,7 @@ static av_cold int g723_1_decode_init(AVCodecContext *avctx)
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* @param buf pointer to the input buffer
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* @param buf_size size of the input buffer
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*/
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static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
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static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf,
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int buf_size)
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{
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GetBitContext gb;
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@ -344,7 +348,7 @@ static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
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* @param ppf pitch postfilter parameters
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* @param cur_rate current bitrate
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*/
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static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
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static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
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PPFParam *ppf, enum Rate cur_rate)
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{
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@ -430,7 +434,7 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
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*
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* @return residual interpolation index if voiced, 0 otherwise
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*/
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static int comp_interp_index(G723_1_Context *p, int pitch_lag,
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static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
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int *exc_eng, int *scale)
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{
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int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
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@ -529,7 +533,7 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag,
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* @param buf postfiltered output vector
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* @param energy input energy coefficient
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*/
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static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
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static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
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{
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int num, denom, gain, bits1, bits2;
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int i;
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@ -572,7 +576,7 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
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* @param buf input buffer
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* @param dst output buffer
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*/
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static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
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static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
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int16_t *buf, int16_t *dst)
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{
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int16_t filter_coef[2][LPC_ORDER];
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@ -655,7 +659,7 @@ static inline int cng_rand(int *state, int base)
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return (*state & 0x7FFF) * base >> 15;
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}
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static int estimate_sid_gain(G723_1_Context *p)
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static int estimate_sid_gain(G723_1_ChannelContext *p)
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{
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int i, shift, seg, seg2, t, val, val_add, x, y;
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@ -715,7 +719,7 @@ static int estimate_sid_gain(G723_1_Context *p)
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return val;
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}
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static void generate_noise(G723_1_Context *p)
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static void generate_noise(G723_1_ChannelContext *p)
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{
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int i, j, idx, t;
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int off[SUBFRAMES];
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@ -843,7 +847,7 @@ static void generate_noise(G723_1_Context *p)
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static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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G723_1_Context *p = avctx->priv_data;
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G723_1_Context *s = avctx->priv_data;
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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@ -855,9 +859,8 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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int16_t acb_vector[SUBFRAME_LEN];
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int16_t *out;
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int bad_frame = 0, i, j, ret;
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int16_t *audio = p->audio;
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if (buf_size < frame_size[dec_mode]) {
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if (buf_size < frame_size[dec_mode] * avctx->channels) {
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if (buf_size)
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av_log(avctx, AV_LOG_WARNING,
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"Expected %d bytes, got %d - skipping packet\n",
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@ -866,6 +869,14 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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return buf_size;
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}
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frame->nb_samples = FRAME_LEN;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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for (int ch = 0; ch < avctx->channels; ch++) {
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G723_1_ChannelContext *p = &s->ch[ch];
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int16_t *audio = p->audio;
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if (unpack_bitstream(p, buf, buf_size) < 0) {
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bad_frame = 1;
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if (p->past_frame_type == ACTIVE_FRAME)
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@ -874,11 +885,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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p->cur_frame_type = UNTRANSMITTED_FRAME;
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}
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frame->nb_samples = FRAME_LEN;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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out = (int16_t *)frame->data[0];
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out = (int16_t *)frame->extended_data[ch];
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if (p->cur_frame_type == ACTIVE_FRAME) {
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if (!bad_frame)
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@ -922,7 +929,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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&p->sid_gain, &p->cur_gain);
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/* Perform pitch postfiltering */
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if (p->postfilter) {
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if (s->postfilter) {
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i = PITCH_MAX;
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for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
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comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
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@ -992,16 +999,17 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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0, 1, 1 << 12);
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memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
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if (p->postfilter) {
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if (s->postfilter) {
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formant_postfilter(p, lpc, p->audio, out);
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} else { // if output is not postfiltered it should be scaled by 2
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for (i = 0; i < FRAME_LEN; i++)
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out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
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}
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}
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*got_frame_ptr = 1;
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return frame_size[dec_mode];
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return frame_size[dec_mode] * avctx->channels;
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}
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#define OFFSET(x) offsetof(G723_1_Context, x)
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@ -42,7 +42,8 @@
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static av_cold int g723_1_encode_init(AVCodecContext *avctx)
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{
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G723_1_Context *p = avctx->priv_data;
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G723_1_Context *s = avctx->priv_data;
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G723_1_ChannelContext *p = &s->ch[0];
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if (avctx->sample_rate != 8000) {
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av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
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@ -386,7 +387,7 @@ static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
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* @param flt_coef filter coefficients
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* @param unq_lpc unquantized lpc vector
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*/
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static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
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static void perceptual_filter(G723_1_ChannelContext *p, int16_t *flt_coef,
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int16_t *unq_lpc, int16_t *buf)
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{
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int16_t vector[FRAME_LEN + LPC_ORDER];
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@ -635,7 +636,7 @@ static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
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* @param buf input signal
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* @param index the current subframe index
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*/
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static void acb_search(G723_1_Context *p, int16_t *residual,
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static void acb_search(G723_1_ChannelContext *p, int16_t *residual,
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int16_t *impulse_resp, const int16_t *buf,
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int index)
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{
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@ -963,7 +964,7 @@ static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
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* @param buf target vector
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* @param impulse_resp impulse response of the combined filter
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*/
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static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
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static void fcb_search(G723_1_ChannelContext *p, int16_t *impulse_resp,
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int16_t *buf, int index)
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{
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FCBParam optim;
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@ -995,7 +996,7 @@ static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
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* @param frame output buffer
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* @param size size of the buffer
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*/
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static int pack_bitstream(G723_1_Context *p, AVPacket *avpkt)
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static int pack_bitstream(G723_1_ChannelContext *p, AVPacket *avpkt)
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{
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PutBitContext pb;
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int info_bits = 0;
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@ -1056,7 +1057,8 @@ static int pack_bitstream(G723_1_Context *p, AVPacket *avpkt)
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static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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G723_1_Context *p = avctx->priv_data;
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G723_1_Context *s = avctx->priv_data;
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G723_1_ChannelContext *p = &s->ch[0];
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int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
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int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
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int16_t cur_lsp[LPC_ORDER];
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