[FFmpeg-cvslog] Add the G723.1 demuxer and decoder
Mohamed Naufal Basheer
git at videolan.org
Thu Sep 29 22:09:47 CEST 2011
ffmpeg | branch: master | Mohamed Naufal Basheer <naufal11 at gmail.com> | Thu Mar 17 23:56:50 2011 +0100| [f990dc374eee84e7e1fd4da1309b9de0bd4f3216] | committer: Michael Niedermayer
Add the G723.1 demuxer and decoder
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f990dc374eee84e7e1fd4da1309b9de0bd4f3216
---
libavcodec/Makefile | 2 +
libavcodec/allcodecs.c | 1 +
libavcodec/avcodec.h | 1 +
libavcodec/g723_1.c | 1081 ++++++++++++++++++++++++++++++++++++++++++++++
libavcodec/version.h | 2 +-
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/g723_1.c | 83 ++++
libavformat/rtp.c | 2 +-
9 files changed, 1172 insertions(+), 2 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 2aa0ffc..27345d9 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -159,6 +159,8 @@ OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o
OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
+OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o acelp_vectors.o \
+ celp_filters.o celp_math.o
OBJS-$(CONFIG_G729_DECODER) += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 72a32c4..03cc807 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -258,6 +258,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_ENCDEC (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
+ REGISTER_DECODER (G723_1, g723_1);
REGISTER_DECODER (G729, g729);
REGISTER_DECODER (GSM, gsm);
REGISTER_DECODER (GSM_MS, gsm_ms);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index e04513a..cb82ec7 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -345,6 +345,7 @@ enum CodecID {
CODEC_ID_QDMC,
CODEC_ID_CELT,
CODEC_ID_G729 = 0x15800,
+ CODEC_ID_G723_1= 0x15801,
/* subtitle codecs */
CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
new file mode 100644
index 0000000..da4f0d2
--- /dev/null
+++ b/libavcodec/g723_1.c
@@ -0,0 +1,1081 @@
+/*
+ * G.723.1 compatible decoder
+ * Copyright (c) 2006 Benjamin Larsson
+ * Copyright (c) 2010 Mohamed Naufal Basheer
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.723.1 compatible decoder
+ */
+
+#include "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "acelp_vectors.h"
+#include "celp_filters.h"
+#include "celp_math.h"
+#include "lsp.h"
+#include "libavutil/lzo.h"
+#include "g723_1_data.h"
+
+typedef struct g723_1_context {
+ G723_1_Subframe subframe[4];
+ FrameType cur_frame_type;
+ FrameType past_frame_type;
+ Rate cur_rate;
+ uint8_t lsp_index[LSP_BANDS];
+ int pitch_lag[2];
+ int erased_frames;
+
+ int16_t prev_lsp[LPC_ORDER];
+ int16_t prev_excitation[PITCH_MAX];
+ int16_t excitation[PITCH_MAX + FRAME_LEN];
+ int16_t synth_mem[LPC_ORDER];
+ int16_t fir_mem[LPC_ORDER];
+ int iir_mem[LPC_ORDER];
+
+ int random_seed;
+ int interp_index;
+ int interp_gain;
+ int sid_gain;
+ int cur_gain;
+ int reflection_coef;
+ int pf_gain; ///< formant postfilter
+ ///< gain scaling unit memory
+} G723_1_Context;
+
+static av_cold int g723_1_decode_init(AVCodecContext *avctx)
+{
+ G723_1_Context *p = avctx->priv_data;
+
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+ p->pf_gain = 1 << 12;
+ memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
+
+ return 0;
+}
+
+/**
+ * Unpack the frame into parameters.
+ *
+ * @param p the context
+ * @param buf pointer to the input buffer
+ * @param buf_size size of the input buffer
+ */
+static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
+ int buf_size)
+{
+ GetBitContext gb;
+ int ad_cb_len;
+ int temp, info_bits, i;
+
+ init_get_bits(&gb, buf, buf_size * 8);
+
+ /* Extract frame type and rate info */
+ info_bits = get_bits(&gb, 2);
+
+ if (info_bits == 3) {
+ p->cur_frame_type = UntransmittedFrame;
+ return 0;
+ }
+
+ /* Extract 24 bit lsp indices, 8 bit for each band */
+ p->lsp_index[2] = get_bits(&gb, 8);
+ p->lsp_index[1] = get_bits(&gb, 8);
+ p->lsp_index[0] = get_bits(&gb, 8);
+
+ if (info_bits == 2) {
+ p->cur_frame_type = SIDFrame;
+ p->subframe[0].amp_index = get_bits(&gb, 6);
+ return 0;
+ }
+
+ /* Extract the info common to both rates */
+ p->cur_rate = info_bits ? Rate5k3 : Rate6k3;
+ p->cur_frame_type = ActiveFrame;
+
+ p->pitch_lag[0] = get_bits(&gb, 7);
+ if (p->pitch_lag[0] > 123) /* test if forbidden code */
+ return -1;
+ p->pitch_lag[0] += PITCH_MIN;
+ p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
+
+ p->pitch_lag[1] = get_bits(&gb, 7);
+ if (p->pitch_lag[1] > 123)
+ return -1;
+ p->pitch_lag[1] += PITCH_MIN;
+ p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
+ p->subframe[0].ad_cb_lag = 1;
+ p->subframe[2].ad_cb_lag = 1;
+
+ for (i = 0; i < SUBFRAMES; i++) {
+ /* Extract combined gain */
+ temp = get_bits(&gb, 12);
+ ad_cb_len = 170;
+ p->subframe[i].dirac_train = 0;
+ if (p->cur_rate == Rate6k3 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
+ p->subframe[i].dirac_train = temp >> 11;
+ temp &= 0x7ff;
+ ad_cb_len = 85;
+ }
+ p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
+ if (p->subframe[i].ad_cb_gain < ad_cb_len) {
+ p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
+ GAIN_LEVELS;
+ } else {
+ return -1;
+ }
+ }
+
+ p->subframe[0].grid_index = get_bits(&gb, 1);
+ p->subframe[1].grid_index = get_bits(&gb, 1);
+ p->subframe[2].grid_index = get_bits(&gb, 1);
+ p->subframe[3].grid_index = get_bits(&gb, 1);
+
+ if (p->cur_rate == Rate6k3) {
+ skip_bits(&gb, 1); /* skip reserved bit */
+
+ /* Compute pulse_pos index using the 13-bit combined position index */
+ temp = get_bits(&gb, 13);
+ p->subframe[0].pulse_pos = temp / 810;
+
+ temp -= p->subframe[0].pulse_pos * 810;
+ p->subframe[1].pulse_pos = FASTDIV(temp, 90);
+
+ temp -= p->subframe[1].pulse_pos * 90;
+ p->subframe[2].pulse_pos = FASTDIV(temp, 9);
+ p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
+
+ p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
+ get_bits(&gb, 16);
+ p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
+ get_bits(&gb, 14);
+ p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
+ get_bits(&gb, 16);
+ p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
+ get_bits(&gb, 14);
+
+ p->subframe[0].pulse_sign = get_bits(&gb, 6);
+ p->subframe[1].pulse_sign = get_bits(&gb, 5);
+ p->subframe[2].pulse_sign = get_bits(&gb, 6);
+ p->subframe[3].pulse_sign = get_bits(&gb, 5);
+ } else { /* Rate5k3 */
+ p->subframe[0].pulse_pos = get_bits(&gb, 12);
+ p->subframe[1].pulse_pos = get_bits(&gb, 12);
+ p->subframe[2].pulse_pos = get_bits(&gb, 12);
+ p->subframe[3].pulse_pos = get_bits(&gb, 12);
+
+ p->subframe[0].pulse_sign = get_bits(&gb, 4);
+ p->subframe[1].pulse_sign = get_bits(&gb, 4);
+ p->subframe[2].pulse_sign = get_bits(&gb, 4);
+ p->subframe[3].pulse_sign = get_bits(&gb, 4);
+ }
+
+ return 0;
+}
+
+/**
+ * Bitexact implementation of sqrt(val/2).
+ */
+static int16_t square_root(int val)
+{
+ int16_t res = 0;
+ int16_t exp = 0x4000;
+ int i;
+
+ for (i = 0; i < 14; i ++) {
+ int res_exp = res + exp;
+ if (val >= res_exp * res_exp << 1)
+ res += exp;
+ exp >>= 1;
+ }
+ return res;
+}
+
+/**
+ * Calculate the number of left-shifts required for normalizing the input.
+ *
+ * @param num input number
+ * @param width width of the input, 16 bits(0) / 32 bits(1)
+ */
+static int normalize_bits(int num, int width)
+{
+ int i = 0;
+ int bits = (width) ? 31 : 15;
+ int limit = 1 << (bits - 1);
+
+ if (num) {
+ if (num == -1)
+ return bits;
+ if (num < 0)
+ num = ~num;
+ for (i = 0; num < limit; i++)
+ num <<= 1;
+ }
+ return i;
+}
+
+/**
+ * Scale vector contents based on the largest of their absolutes.
+ */
+static int scale_vector(int16_t *vector, int length)
+{
+ int bits, scale, max = 0;
+ int i;
+
+ const int16_t shift_table[16] = {
+ 0x0001, 0x0002, 0x0004, 0x0008, 0x0010, 0x0020, 0x0040, 0x0080,
+ 0x0100, 0x0200, 0x0400, 0x0800, 0x1000, 0x2000, 0x4000, 0x7fff
+ };
+
+ for (i = 0; i < length; i++)
+ max = FFMAX(max, FFABS(vector[i]));
+
+ bits = normalize_bits(max, 0);
+ scale = shift_table[bits];
+
+ for (i = 0; i < length; i++)
+ vector[i] = (int16_t)(av_clipl_int32(vector[i] * scale << 1) >> 4);
+
+ return bits - 3;
+}
+
+/**
+ * Perform inverse quantization of LSP frequencies.
+ *
+ * @param cur_lsp the current LSP vector
+ * @param prev_lsp the previous LSP vector
+ * @param lsp_index VQ indices
+ * @param bad_frame bad frame flag
+ */
+static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
+ uint8_t *lsp_index, int bad_frame)
+{
+ int min_dist, pred;
+ int i, j, temp, stable;
+
+ /* Check for frame erasure */
+ if (!bad_frame) {
+ min_dist = 0x100;
+ pred = 12288;
+ } else {
+ min_dist = 0x200;
+ pred = 23552;
+ lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
+ }
+
+ /* Get the VQ table entry corresponding to the transmitted index */
+ cur_lsp[0] = lsp_band0[lsp_index[0]][0];
+ cur_lsp[1] = lsp_band0[lsp_index[0]][1];
+ cur_lsp[2] = lsp_band0[lsp_index[0]][2];
+ cur_lsp[3] = lsp_band1[lsp_index[1]][0];
+ cur_lsp[4] = lsp_band1[lsp_index[1]][1];
+ cur_lsp[5] = lsp_band1[lsp_index[1]][2];
+ cur_lsp[6] = lsp_band2[lsp_index[2]][0];
+ cur_lsp[7] = lsp_band2[lsp_index[2]][1];
+ cur_lsp[8] = lsp_band2[lsp_index[2]][2];
+ cur_lsp[9] = lsp_band2[lsp_index[2]][3];
+
+ /* Add predicted vector & DC component to the previously quantized vector */
+ for (i = 0; i < LPC_ORDER; i++) {
+ temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
+ cur_lsp[i] += dc_lsp[i] + temp;
+ }
+
+ for (i = 0; i < LPC_ORDER; i++) {
+ cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
+ cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
+
+ /* Stability check */
+ for (j = 1; j < LPC_ORDER; j++) {
+ temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
+ if (temp > 0) {
+ temp >>= 1;
+ cur_lsp[j - 1] -= temp;
+ cur_lsp[j] += temp;
+ }
+ }
+ stable = 1;
+ for (j = 1; j < LPC_ORDER; j++) {
+ temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
+ if (temp > 0) {
+ stable = 0;
+ break;
+ }
+ }
+ if (stable)
+ break;
+ }
+ if (!stable)
+ memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
+}
+
+/**
+ * Bitexact implementation of 2ab scaled by 1/2^16.
+ *
+ * @param a 32 bit multiplicand
+ * @param b 16 bit multiplier
+ */
+#define MULL2(a, b) \
+ ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))
+
+/**
+ * Convert LSP frequencies to LPC coefficients.
+ *
+ * @param lpc buffer for LPC coefficients
+ */
+static void lsp2lpc(int16_t *lpc)
+{
+ int f1[LPC_ORDER / 2 + 1];
+ int f2[LPC_ORDER / 2 + 1];
+ int i, j;
+
+ /* Calculate negative cosine */
+ for (j = 0; j < LPC_ORDER; j++) {
+ int index = lpc[j] >> 7;
+ int offset = lpc[j] & 0x7f;
+ int64_t temp1 = cos_tab[index] << 16;
+ int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
+ ((offset << 8) + 0x80) << 1;
+
+ lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16);
+ }
+
+ /*
+ * Compute sum and difference polynomial coefficients
+ * (bitexact alternative to lsp2poly() in lsp.c)
+ */
+ /* Initialize with values in Q28 */
+ f1[0] = 1 << 28;
+ f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
+ f1[2] = lpc[0] * lpc[2] + (2 << 28);
+
+ f2[0] = 1 << 28;
+ f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
+ f2[2] = lpc[1] * lpc[3] + (2 << 28);
+
+ /*
+ * Calculate and scale the coefficients by 1/2 in
+ * each iteration for a final scaling factor of Q25
+ */
+ for (i = 2; i < LPC_ORDER / 2; i++) {
+ f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
+ f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
+
+ for (j = i; j >= 2; j--) {
+ f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
+ (f1[j] >> 1) + (f1[j - 2] >> 1);
+ f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
+ (f2[j] >> 1) + (f2[j - 2] >> 1);
+ }
+
+ f1[0] >>= 1;
+ f2[0] >>= 1;
+ f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;
+ f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
+ }
+
+ /* Convert polynomial coefficients to LPC coefficients */
+ for (i = 0; i < LPC_ORDER / 2; i++) {
+ int64_t ff1 = f1[i + 1] + f1[i];
+ int64_t ff2 = f2[i + 1] - f2[i];
+
+ lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
+ lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
+ (1 << 15)) >> 16;
+ }
+}
+
+/**
+ * Quantize LSP frequencies by interpolation and convert them to
+ * the corresponding LPC coefficients.
+ *
+ * @param lpc buffer for LPC coefficients
+ * @param cur_lsp the current LSP vector
+ * @param prev_lsp the previous LSP vector
+ */
+static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
+{
+ int i;
+ int16_t *lpc_ptr = lpc;
+
+ /* cur_lsp * 0.25 + prev_lsp * 0.75 */
+ ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
+ 4096, 12288, 1 << 13, 14, LPC_ORDER);
+ ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
+ 8192, 8192, 1 << 13, 14, LPC_ORDER);
+ ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
+ 12288, 4096, 1 << 13, 14, LPC_ORDER);
+ memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(int16_t));
+
+ for (i = 0; i < SUBFRAMES; i++) {
+ lsp2lpc(lpc_ptr);
+ lpc_ptr += LPC_ORDER;
+ }
+}
+
+/**
+ * Generate a train of dirac functions with period as pitch lag.
+ */
+static void gen_dirac_train(int16_t *buf, int pitch_lag)
+{
+ int16_t vector[SUBFRAME_LEN];
+ int i, j;
+
+ memcpy(vector, buf, SUBFRAME_LEN * sizeof(int16_t));
+ for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
+ for (j = 0; j < SUBFRAME_LEN - i; j++)
+ buf[i + j] += vector[j];
+ }
+}
+
+/**
+ * Generate fixed codebook excitation vector.
+ *
+ * @param vector decoded excitation vector
+ * @param subfrm current subframe
+ * @param cur_rate current bitrate
+ * @param pitch_lag closed loop pitch lag
+ * @param index current subframe index
+ */
+static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
+ Rate cur_rate, int pitch_lag, int index)
+{
+ int temp, i, j;
+
+ memset(vector, 0, SUBFRAME_LEN * sizeof(int16_t));
+
+ if (cur_rate == Rate6k3) {
+ if (subfrm.pulse_pos >= max_pos[index])
+ return;
+
+ /* Decode amplitudes and positions */
+ j = PULSE_MAX - pulses[index];
+ temp = subfrm.pulse_pos;
+ for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
+ temp -= combinatorial_table[j][i];
+ if (temp >= 0)
+ continue;
+ temp += combinatorial_table[j++][i];
+ if (subfrm.pulse_sign & (1 << (PULSE_MAX - j))) {
+ vector[subfrm.grid_index + GRID_SIZE * i] =
+ -fixed_cb_gain[subfrm.amp_index];
+ } else {
+ vector[subfrm.grid_index + GRID_SIZE * i] =
+ fixed_cb_gain[subfrm.amp_index];
+ }
+ if (j == PULSE_MAX)
+ break;
+ }
+ if (subfrm.dirac_train == 1)
+ gen_dirac_train(vector, pitch_lag);
+ } else { /* Rate5k3 */
+ int cb_gain = fixed_cb_gain[subfrm.amp_index];
+ int cb_shift = subfrm.grid_index;
+ int cb_sign = subfrm.pulse_sign;
+ int cb_pos = subfrm.pulse_pos;
+ int offset, beta, lag;
+
+ for (i = 0; i < 8; i += 2) {
+ offset = ((cb_pos & 7) << 3) + cb_shift + i;
+ vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
+ cb_pos >>= 3;
+ cb_sign >>= 1;
+ }
+
+ /* Enhance harmonic components */
+ lag = pitch_contrib[subfrm.ad_cb_gain << 1] + pitch_lag +
+ subfrm.ad_cb_lag - 1;
+ beta = pitch_contrib[(subfrm.ad_cb_gain << 1) + 1];
+
+ if (lag < SUBFRAME_LEN - 2) {
+ for (i = lag; i < SUBFRAME_LEN; i++)
+ vector[i] += beta * vector[i - lag] >> 15;
+ }
+ }
+}
+
+/**
+ * Get delayed contribution from the previous excitation vector.
+ */
+static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
+{
+ int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
+ int i;
+
+ residual[0] = prev_excitation[offset];
+ residual[1] = prev_excitation[offset + 1];
+
+ offset += 2;
+ for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
+ residual[i] = prev_excitation[offset + (i - 2) % lag];
+}
+
+/**
+ * Generate adaptive codebook excitation.
+ */
+static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
+ int pitch_lag, G723_1_Subframe subfrm,
+ Rate cur_rate)
+{
+ int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
+ const int16_t *cb_ptr;
+ int lag = pitch_lag + subfrm.ad_cb_lag - 1;
+
+ int i;
+ int64_t sum;
+
+ get_residual(residual, prev_excitation, lag);
+
+ /* Select quantization table */
+ if (cur_rate == Rate6k3 && pitch_lag < SUBFRAME_LEN - 2) {
+ cb_ptr = adaptive_cb_gain85;
+ } else
+ cb_ptr = adaptive_cb_gain170;
+
+ /* Calculate adaptive vector */
+ cb_ptr += subfrm.ad_cb_gain * 20;
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER, 1);
+ vector[i] = av_clipl_int32((sum << 1) + (1 << 15)) >> 16;
+ }
+}
+
+/**
+ * Estimate maximum auto-correlation around pitch lag.
+ *
+ * @param p the context
+ * @param offset offset of the excitation vector
+ * @param ccr_max pointer to the maximum auto-correlation
+ * @param pitch_lag decoded pitch lag
+ * @param length length of autocorrelation
+ * @param dir forward lag(1) / backward lag(-1)
+ */
+static int autocorr_max(G723_1_Context *p, int offset, int *ccr_max,
+ int pitch_lag, int length, int dir)
+{
+ int limit, ccr, lag = 0;
+ int16_t *buf = p->excitation + offset;
+ int i;
+
+ pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
+ limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
+
+ for (i = pitch_lag - 3; i <= limit; i++) {
+ ccr = ff_dot_product(buf, buf + dir * i, length, 1);
+
+ if (ccr > *ccr_max) {
+ *ccr_max = ccr;
+ lag = i;
+ }
+ }
+ return lag;
+}
+
+/**
+ * Calculate pitch postfilter optimal and scaling gains.
+ *
+ * @param lag pitch postfilter forward/backward lag
+ * @param ppf pitch postfilter parameters
+ * @param cur_rate current bitrate
+ * @param tgt_eng target energy
+ * @param ccr cross-correlation
+ * @param res_eng residual energy
+ */
+static void comp_ppf_gains(int lag, PPFParam *ppf, Rate cur_rate,
+ int tgt_eng, int ccr, int res_eng)
+{
+ int pf_residual; /* square of postfiltered residual */
+ int64_t temp1, temp2;
+
+ ppf->index = lag;
+
+ temp1 = tgt_eng * res_eng >> 1;
+ temp2 = ccr * ccr << 1;
+
+ if (temp2 > temp1) {
+ if (ccr >= res_eng) {
+ ppf->opt_gain = ppf_gain_weight[cur_rate];
+ } else {
+ ppf->opt_gain = (ccr << 15) / res_eng *
+ ppf_gain_weight[cur_rate] >> 15;
+ }
+ /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
+ temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
+ temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
+ pf_residual = av_clipl_int32(temp1 + temp2 + (1 << 15)) >> 16;
+
+ if (tgt_eng >= pf_residual << 1) {
+ temp1 = 0x7fff;
+ } else {
+ temp1 = (tgt_eng << 14) / pf_residual;
+ }
+
+ /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
+ ppf->sc_gain = square_root(temp1 << 16);
+ } else {
+ ppf->opt_gain = 0;
+ ppf->sc_gain = 0x7fff;
+ }
+
+ ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
+}
+
+/**
+ * Calculate pitch postfilter parameters.
+ *
+ * @param p the context
+ * @param offset offset of the excitation vector
+ * @param pitch_lag decoded pitch lag
+ * @param ppf pitch postfilter parameters
+ * @param cur_rate current bitrate
+ */
+static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
+ PPFParam *ppf, Rate cur_rate)
+{
+
+ int16_t scale;
+ int i;
+ int64_t temp1, temp2;
+
+ /*
+ * 0 - target energy
+ * 1 - forward cross-correlation
+ * 2 - forward residual energy
+ * 3 - backward cross-correlation
+ * 4 - backward residual energy
+ */
+ int energy[5] = {0, 0, 0, 0, 0};
+ int16_t *buf = p->excitation + offset;
+ int fwd_lag = autocorr_max(p, offset, &energy[1], pitch_lag,
+ SUBFRAME_LEN, 1);
+ int back_lag = autocorr_max(p, offset, &energy[3], pitch_lag,
+ SUBFRAME_LEN, -1);
+
+ ppf->index = 0;
+ ppf->opt_gain = 0;
+ ppf->sc_gain = 0x7fff;
+
+ /* Case 0, Section 3.6 */
+ if (!back_lag && !fwd_lag)
+ return;
+
+ /* Compute target energy */
+ energy[0] = ff_dot_product(buf, buf, SUBFRAME_LEN, 1);
+
+ /* Compute forward residual energy */
+ if (fwd_lag)
+ energy[2] = ff_dot_product(buf + fwd_lag, buf + fwd_lag,
+ SUBFRAME_LEN, 1);
+
+ /* Compute backward residual energy */
+ if (back_lag)
+ energy[4] = ff_dot_product(buf - back_lag, buf - back_lag,
+ SUBFRAME_LEN, 1);
+
+ /* Normalize and shorten */
+ temp1 = 0;
+ for (i = 0; i < 5; i++)
+ temp1 = FFMAX(energy[i], temp1);
+
+ scale = normalize_bits(temp1, 1);
+ for (i = 0; i < 5; i++)
+ energy[i] = av_clipl_int32(energy[i] << scale) >> 16;
+
+ if (fwd_lag && !back_lag) { /* Case 1 */
+ comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
+ energy[2]);
+ } else if (!fwd_lag) { /* Case 2 */
+ comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
+ energy[4]);
+ } else { /* Case 3 */
+
+ /*
+ * Select the largest of energy[1]^2/energy[2]
+ * and energy[3]^2/energy[4]
+ */
+ temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
+ temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
+ if (temp1 >= temp2) {
+ comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
+ energy[2]);
+ } else {
+ comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
+ energy[4]);
+ }
+ }
+}
+
+/**
+ * Classify frames as voiced/unvoiced.
+ *
+ * @param p the context
+ * @param pitch_lag decoded pitch_lag
+ * @param exc_eng excitation energy estimation
+ * @param scale scaling factor of exc_eng
+ *
+ * @return residual interpolation index if voiced, 0 otherwise
+ */
+static int comp_interp_index(G723_1_Context *p, int pitch_lag,
+ int *exc_eng, int *scale)
+{
+ int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
+ int16_t *buf = p->excitation + offset;
+
+ int index, ccr, tgt_eng, best_eng, temp;
+
+ *scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX);
+
+ /* Compute maximum backward cross-correlation */
+ ccr = 0;
+ index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
+ ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16;
+
+ /* Compute target energy */
+ tgt_eng = ff_dot_product(buf, buf, SUBFRAME_LEN * 2, 1);
+ *exc_eng = av_clipl_int32(tgt_eng + (1 << 15)) >> 16;
+
+ if (ccr <= 0)
+ return 0;
+
+ /* Compute best energy */
+ best_eng = ff_dot_product(buf - index, buf - index,
+ SUBFRAME_LEN * 2, 1);
+ best_eng = av_clipl_int32((int64_t)best_eng + (1 << 15)) >> 16;
+
+ temp = best_eng * *exc_eng >> 3;
+
+ if (temp < ccr * ccr) {
+ return index;
+ } else
+ return 0;
+}
+
+/**
+ * Peform residual interpolation based on frame classification.
+ *
+ * @param buf decoded excitation vector
+ * @param out output vector
+ * @param lag decoded pitch lag
+ * @param gain interpolated gain
+ * @param rseed seed for random number generator
+ */
+static void residual_interp(int16_t *buf, int16_t *out, int lag,
+ int gain, int *rseed)
+{
+ int i;
+ if (lag) { /* Voiced */
+ int16_t *vector_ptr = buf + PITCH_MAX;
+ /* Attenuate */
+ for (i = 0; i < lag; i++)
+ vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2;
+ av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(int16_t),
+ FRAME_LEN * sizeof(int16_t));
+ memcpy(out, vector_ptr, FRAME_LEN * sizeof(int16_t));
+ } else { /* Unvoiced */
+ for (i = 0; i < FRAME_LEN; i++) {
+ *rseed = *rseed * 521 + 259;
+ out[i] = gain * *rseed >> 15;
+ }
+ memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(int16_t));
+ }
+}
+
+/**
+ * Perform IIR filtering.
+ *
+ * @param fir_coef FIR coefficients
+ * @param iir_coef IIR coefficients
+ * @param src source vector
+ * @param dest destination vector
+ * @param width width of the output, 16 bits(0) / 32 bits(1)
+ */
+#define iir_filter(fir_coef, iir_coef, src, dest, width)\
+{\
+ int m, n;\
+ int res_shift = 16 & ~-(width);\
+ int in_shift = 16 - res_shift;\
+\
+ for (m = 0; m < SUBFRAME_LEN; m++) {\
+ int64_t filter = 0;\
+ for (n = 1; n <= LPC_ORDER; n++) {\
+ filter -= (fir_coef)[n - 1] * (src)[m - n] -\
+ (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
+ }\
+\
+ (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
+ (1 << 15)) >> res_shift;\
+ }\
+}
+
+/**
+ * Adjust gain of postfiltered signal.
+ *
+ * @param p the context
+ * @param buf postfiltered output vector
+ * @param energy input energy coefficient
+ */
+static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
+{
+ int num, denom, gain, bits1, bits2;
+ int i;
+
+ num = energy;
+ denom = 0;
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t temp = buf[i] >> 2;
+ temp = av_clipl_int32(MUL64(temp, temp) << 1);
+ denom = av_clipl_int32(denom + temp);
+ }
+
+ if (num && denom) {
+ bits1 = normalize_bits(num, 1);
+ bits2 = normalize_bits(denom, 1);
+ num = num << bits1 >> 1;
+ denom <<= bits2;
+
+ bits2 = 5 + bits1 - bits2;
+ bits2 = FFMAX(0, bits2);
+
+ gain = (num >> 1) / (denom >> 16);
+ gain = square_root(gain << 16 >> bits2);
+ } else {
+ gain = 1 << 12;
+ }
+
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ p->pf_gain = ((p->pf_gain << 4) - p->pf_gain + gain + (1 << 3)) >> 4;
+ buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
+ (1 << 10)) >> 11);
+ }
+}
+
+/**
+ * Perform formant filtering.
+ *
+ * @param p the context
+ * @param lpc quantized lpc coefficients
+ * @param buf output buffer
+ */
+static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
+{
+ int16_t filter_coef[2][LPC_ORDER], *buf_ptr;
+ int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
+ int i, j, k;
+
+ memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(int16_t));
+ memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(int));
+
+ for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
+ for (k = 0; k < LPC_ORDER; k++) {
+ filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
+ (1 << 14)) >> 15;
+ filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
+ (1 << 14)) >> 15;
+ }
+ iir_filter(filter_coef[0], filter_coef[1], buf + i,
+ filter_signal + i, 1);
+ }
+
+ memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
+ memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
+
+ buf_ptr = buf + LPC_ORDER;
+ signal_ptr = filter_signal + LPC_ORDER;
+ for (i = 0; i < SUBFRAMES; i++) {
+ int16_t temp_vector[SUBFRAME_LEN];
+ int16_t temp;
+ int auto_corr[2];
+ int scale, energy;
+
+ /* Normalize */
+ memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(int16_t));
+ scale = scale_vector(temp_vector, SUBFRAME_LEN);
+
+ /* Compute auto correlation coefficients */
+ auto_corr[0] = ff_dot_product(temp_vector, temp_vector + 1,
+ SUBFRAME_LEN - 1, 1);
+ auto_corr[1] = ff_dot_product(temp_vector, temp_vector,
+ SUBFRAME_LEN, 1);
+
+ /* Compute reflection coefficient */
+ temp = auto_corr[1] >> 16;
+ if (temp) {
+ temp = (auto_corr[0] >> 2) / temp;
+ }
+ p->reflection_coef = ((p->reflection_coef << 2) - p->reflection_coef +
+ temp + 2) >> 2;
+ temp = (p->reflection_coef * 0xffffc >> 3) & 0xfffc;
+
+ /* Compensation filter */
+ for (j = 0; j < SUBFRAME_LEN; j++) {
+ buf_ptr[j] = av_clipl_int32(signal_ptr[j] +
+ ((signal_ptr[j - 1] >> 16) *
+ temp << 1)) >> 16;
+ }
+
+ /* Compute normalized signal energy */
+ temp = 2 * scale + 4;
+ if (temp < 0) {
+ energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
+ } else
+ energy = auto_corr[1] >> temp;
+
+ gain_scale(p, buf_ptr, energy);
+
+ buf_ptr += SUBFRAME_LEN;
+ signal_ptr += SUBFRAME_LEN;
+ }
+}
+
+static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
+ int *data_size, AVPacket *avpkt)
+{
+ G723_1_Context *p = avctx->priv_data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ int16_t *out = data;
+ int dec_mode = buf[0] & 3;
+
+ PPFParam ppf[SUBFRAMES];
+ int16_t cur_lsp[LPC_ORDER];
+ int16_t lpc[SUBFRAMES * LPC_ORDER];
+ int16_t acb_vector[SUBFRAME_LEN];
+ int16_t *vector_ptr;
+ int bad_frame = 0, i, j;
+
+ if (!buf_size || buf_size < frame_size[dec_mode]) {
+ *data_size = 0;
+ return buf_size;
+ }
+
+ if (unpack_bitstream(p, buf, buf_size) < 0) {
+ bad_frame = 1;
+ p->cur_frame_type = p->past_frame_type == ActiveFrame ?
+ ActiveFrame : UntransmittedFrame;
+ }
+
+ *data_size = FRAME_LEN * sizeof(int16_t);
+ if(p->cur_frame_type == ActiveFrame) {
+ if (!bad_frame) {
+ p->erased_frames = 0;
+ } else if(p->erased_frames != 3)
+ p->erased_frames++;
+
+ inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
+ lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
+
+ /* Save the lsp_vector for the next frame */
+ memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(int16_t));
+
+ /* Generate the excitation for the frame */
+ memcpy(p->excitation, p->prev_excitation, PITCH_MAX * sizeof(int16_t));
+ vector_ptr = p->excitation + PITCH_MAX;
+ if (!p->erased_frames) {
+ /* Update interpolation gain memory */
+ p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
+ p->subframe[3].amp_index) >> 1];
+ for (i = 0; i < SUBFRAMES; i++) {
+ gen_fcb_excitation(vector_ptr, p->subframe[i], p->cur_rate,
+ p->pitch_lag[i >> 1], i);
+ gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
+ p->pitch_lag[i >> 1], p->subframe[i],
+ p->cur_rate);
+ /* Get the total excitation */
+ for (j = 0; j < SUBFRAME_LEN; j++) {
+ vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1);
+ vector_ptr[j] = av_clip_int16(vector_ptr[j] +
+ acb_vector[j]);
+ }
+ vector_ptr += SUBFRAME_LEN;
+ }
+
+ vector_ptr = p->excitation + PITCH_MAX;
+
+ /* Save the excitation */
+ memcpy(out, vector_ptr, FRAME_LEN * sizeof(int16_t));
+
+ p->interp_index = comp_interp_index(p, p->pitch_lag[1],
+ &p->sid_gain, &p->cur_gain);
+
+ for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
+ ppf + j, p->cur_rate);
+
+ /* Restore the original excitation */
+ memcpy(p->excitation, p->prev_excitation,
+ PITCH_MAX * sizeof(int16_t));
+ memcpy(vector_ptr, out, FRAME_LEN * sizeof(int16_t));
+
+ /* Peform pitch postfiltering */
+ for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_acelp_weighted_vector_sum(out + LPC_ORDER + i, vector_ptr + i,
+ vector_ptr + i + ppf[j].index,
+ ppf[j].sc_gain, ppf[j].opt_gain,
+ 1 << 14, 15, SUBFRAME_LEN);
+ } else {
+ p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
+ if (p->erased_frames == 3) {
+ /* Mute output */
+ memset(p->excitation, 0,
+ (FRAME_LEN + PITCH_MAX) * sizeof(int16_t));
+ memset(out, 0, (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
+ } else {
+ /* Regenerate frame */
+ residual_interp(p->excitation, out + LPC_ORDER, p->interp_index,
+ p->interp_gain, &p->random_seed);
+ }
+ }
+ /* Save the excitation for the next frame */
+ memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
+ PITCH_MAX * sizeof(int16_t));
+ } else {
+ memset(out, 0, *data_size);
+ av_log(avctx, AV_LOG_WARNING,
+ "G.723.1: Comfort noise generation not supported yet\n");
+ return frame_size[dec_mode];
+ }
+
+ p->past_frame_type = p->cur_frame_type;
+
+ memcpy(out, p->synth_mem, LPC_ORDER * sizeof(int16_t));
+ for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_celp_lp_synthesis_filter(out + i, &lpc[j * LPC_ORDER],
+ out + i, SUBFRAME_LEN, LPC_ORDER,
+ 0, 1, 1 << 12);
+ memcpy(p->synth_mem, out + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
+
+ formant_postfilter(p, lpc, out);
+
+ memmove(out, out + LPC_ORDER, *data_size);
+
+ return frame_size[dec_mode];
+}
+
+AVCodec ff_g723_1_decoder = {
+ .name = "g723_1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_G723_1,
+ .priv_data_size = sizeof(G723_1_Context),
+ .init = g723_1_decode_init,
+ .decode = g723_1_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
+ .capabilities = CODEC_CAP_SUBFRAMES,
+};
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 872857d..1b5ee01 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -21,7 +21,7 @@
#define AVCODEC_VERSION_H
#define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR 18
+#define LIBAVCODEC_VERSION_MINOR 19
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 04e995f..5f3bf4d 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -98,6 +98,7 @@ OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o
OBJS-$(CONFIG_G722_DEMUXER) += rawdec.o
OBJS-$(CONFIG_G722_MUXER) += rawenc.o
+OBJS-$(CONFIG_G723_1_DEMUXER) += g723_1.o
OBJS-$(CONFIG_H261_DEMUXER) += h261dec.o rawdec.o
OBJS-$(CONFIG_H261_MUXER) += rawenc.o
OBJS-$(CONFIG_H263_DEMUXER) += h263dec.o rawdec.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 429ccc5..94421ab 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -100,6 +100,7 @@ void av_register_all(void)
REGISTER_MUXER (FRAMECRC, framecrc);
REGISTER_MUXER (FRAMEMD5, framemd5);
REGISTER_MUXDEMUX (G722, g722);
+ REGISTER_DEMUXER (G723_1, g723_1);
REGISTER_MUXER (GIF, gif);
REGISTER_DEMUXER (GSM, gsm);
REGISTER_MUXDEMUX (GXF, gxf);
diff --git a/libavformat/g723_1.c b/libavformat/g723_1.c
new file mode 100644
index 0000000..5ffd7c7
--- /dev/null
+++ b/libavformat/g723_1.c
@@ -0,0 +1,83 @@
+/*
+ * G.723.1 demuxer
+ * Copyright (c) 2010 Mohamed Naufal Basheer
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.723.1 demuxer
+ */
+
+#include "avformat.h"
+
+static const uint8_t frame_size[4] = {24, 20, 4, 1};
+
+static int g723_1_init(AVFormatContext *s, AVFormatParameters *ap)
+{
+ AVStream *st;
+
+ st = av_new_stream(s, 0);
+ if (!st)
+ return AVERROR(ENOMEM);
+
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->codec_id = CODEC_ID_G723_1;
+ st->codec->channels = 1;
+ st->codec->sample_rate = 8000;
+
+ av_set_pts_info(st, 64, 1, st->codec->sample_rate);
+
+ return 0;
+}
+
+static int g723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ int size, byte, ret;
+
+ pkt->pos = url_ftell(s->pb);
+ byte = get_byte(s->pb);
+ size = frame_size[byte & 3];
+
+ ret = av_new_packet(pkt, size);
+ if (ret < 0)
+ return ret;
+
+ pkt->data[0] = byte;
+ pkt->duration = 240;
+ pkt->stream_index = 0;
+
+ ret = get_buffer(s->pb, pkt->data + 1, size - 1);
+ if (ret < size - 1) {
+ av_free_packet(pkt);
+ return ret < 0 ? ret : AVERROR_EOF;
+ }
+
+ return pkt->size;
+}
+
+AVInputFormat ff_g723_1_demuxer = {
+ "g723_1",
+ NULL_IF_CONFIG_SMALL("G.723.1 format"),
+ 0,
+ NULL,
+ g723_1_init,
+ g723_1_read_packet,
+ .extensions = "tco",
+ .flags = AVFMT_GENERIC_INDEX
+};
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 6028fe0..127b540 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -44,7 +44,7 @@ static const struct
{
{0, "PCMU", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
{3, "GSM", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
- {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_G723_1, 8000, 1},
{5, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{6, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
{7, "LPC", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
More information about the ffmpeg-cvslog
mailing list