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-rw-r--r--libavcodec/amrnbdec.c78
1 files changed, 48 insertions, 30 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index 7692cf01c6..43ddb625eb 100644
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -3,20 +3,20 @@
* Copyright (c) 2006-2007 Robert Swain
* Copyright (c) 2009 Colin McQuillan
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -47,6 +47,8 @@
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "libavutil/common.h"
+#include "libavutil/avassert.h"
+#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
@@ -84,7 +86,7 @@
/** Maximum sharpening factor
*
* The specification says 0.8, which should be 13107, but the reference C code
- * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in bitexact G.729.)
+ * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
*/
#define SHARP_MAX 0.79449462890625
@@ -136,6 +138,11 @@ typedef struct AMRContext {
float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
+ ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
+ ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
+ CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
+ CELPMContext celpm_ctx; ///< context for fixed point math operations
+
} AMRContext;
/** Double version of ff_weighted_vector_sumf() */
@@ -162,7 +169,8 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
- avctx->sample_rate = 8000;
+ if (!avctx->sample_rate)
+ avctx->sample_rate = 8000;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
// p->excitation always points to the same position in p->excitation_buf
@@ -176,6 +184,11 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
for (i = 0; i < 4; i++)
p->prediction_error[i] = MIN_ENERGY;
+ ff_acelp_filter_init(&p->acelpf_ctx);
+ ff_acelp_vectors_init(&p->acelpv_ctx);
+ ff_celp_filter_init(&p->celpf_ctx);
+ ff_celp_math_init(&p->celpm_ctx);
+
return 0;
}
@@ -219,15 +232,16 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
* Interpolate the LSF vector (used for fixed gain smoothing).
* The interpolation is done over all four subframes even in MODE_12k2.
*
+ * @param[in] ctx The Context
* @param[in,out] lsf_q LSFs in [0,1] for each subframe
* @param[in] lsf_new New LSFs in [0,1] for subframe 4
*/
-static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
+static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
{
int i;
for (i = 0; i < 4; i++)
- ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
+ ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
0.25 * (3 - i), 0.25 * (i + 1),
LP_FILTER_ORDER);
}
@@ -271,7 +285,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
if (update)
- interpolate_lsf(p->lsf_q, lsf_q);
+ interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
}
@@ -334,7 +348,7 @@ static void lsf2lsp_3(AMRContext *p)
ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
// store data for computing the next frame's LSFs
- interpolate_lsf(p->lsf_q, lsf_q);
+ interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
@@ -400,7 +414,8 @@ static void decode_pitch_vector(AMRContext *p,
/* Calculate the pitch vector by interpolating the past excitation at the
pitch lag using a b60 hamming windowed sinc function. */
- ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
+ p->acelpf_ctx.acelp_interpolatef(p->excitation,
+ p->excitation + 1 - pitch_lag_int,
ff_b60_sinc, 6,
pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
10, AMR_SUBFRAME_SIZE);
@@ -484,7 +499,7 @@ static void decode_8_pulses_31bits(const int16_t *fixed_index,
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
const enum Mode mode, const int subframe)
{
- assert(MODE_4k75 <= mode && mode <= MODE_12k2);
+ av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);
if (mode == MODE_12k2) {
ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
@@ -785,12 +800,12 @@ static int synthesis(AMRContext *p, float *lpc,
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
p->pitch_vector[i] *= 0.25;
- ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
+ p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
// emphasize pitch vector contribution
if (p->pitch_gain[4] > 0.5 && !overflow) {
- float energy = avpriv_scalarproduct_float_c(excitation, excitation,
+ float energy = p->celpm_ctx.dot_productf(excitation, excitation,
AMR_SUBFRAME_SIZE);
float pitch_factor =
p->pitch_gain[4] *
@@ -805,7 +820,8 @@ static int synthesis(AMRContext *p, float *lpc,
AMR_SUBFRAME_SIZE);
}
- ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
+ p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
+ AMR_SUBFRAME_SIZE,
LP_FILTER_ORDER);
// detect overflow
@@ -851,10 +867,11 @@ static void update_state(AMRContext *p)
/**
* Get the tilt factor of a formant filter from its transfer function
*
+ * @param p The Context
* @param lpc_n LP_FILTER_ORDER coefficients of the numerator
* @param lpc_d LP_FILTER_ORDER coefficients of the denominator
*/
-static float tilt_factor(float *lpc_n, float *lpc_d)
+static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
{
float rh0, rh1; // autocorrelation at lag 0 and 1
@@ -864,11 +881,12 @@ static float tilt_factor(float *lpc_n, float *lpc_d)
hf[0] = 1.0;
memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
- ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
+ p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
+ AMR_TILT_RESPONSE,
LP_FILTER_ORDER);
- rh0 = avpriv_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE);
- rh1 = avpriv_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1);
+ rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
+ rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
// The spec only specifies this check for 12.2 and 10.2 kbit/s
// modes. But in the ref source the tilt is always non-negative.
@@ -888,7 +906,7 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
int i;
float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
- float speech_gain = avpriv_scalarproduct_float_c(samples, samples,
+ float speech_gain = p->celpm_ctx.dot_productf(samples, samples,
AMR_SUBFRAME_SIZE);
float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
@@ -909,16 +927,16 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
}
memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
- ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
+ p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
sizeof(float) * LP_FILTER_ORDER);
- ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
+ p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
pole_out + LP_FILTER_ORDER,
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
- ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
+ ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
AMR_SUBFRAME_SIZE);
ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
@@ -945,10 +963,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
/* get output buffer */
frame->nb_samples = AMR_BLOCK_SIZE;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- }
buf_out = (float *)frame->data[0];
p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
@@ -957,7 +973,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
if (p->cur_frame_mode == MODE_DTX) {
- avpriv_request_sample(avctx, "dtx mode");
+ avpriv_report_missing_feature(avctx, "dtx mode");
+ av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
return AVERROR_PATCHWELCOME;
}
@@ -995,7 +1012,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
p->fixed_gain[4] =
ff_amr_set_fixed_gain(fixed_gain_factor,
- avpriv_scalarproduct_float_c(p->fixed_vector,
+ p->celpm_ctx.dot_productf(p->fixed_vector,
p->fixed_vector,
AMR_SUBFRAME_SIZE) /
AMR_SUBFRAME_SIZE,
@@ -1041,7 +1058,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
update_state(p);
}
- ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
+ p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
+ buf_out, highpass_zeros,
highpass_poles,
highpass_gain * AMR_SAMPLE_SCALE,
p->high_pass_mem, AMR_BLOCK_SIZE);
@@ -1052,7 +1070,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
* for fixed_gain_smooth.
* The specification has an incorrect formula: the reference decoder uses
* qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
- ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
+ p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
0.84, 0.16, LP_FILTER_ORDER);
*got_frame_ptr = 1;