Opus integration
First patch = delivery from August 22, 2012.
Review URL: https://webrtc-codereview.appspot.com/756005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2945 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/DEPS b/DEPS
index 46c3d2c..224e94b 100644
--- a/DEPS
+++ b/DEPS
@@ -57,6 +57,9 @@
"third_party/libyuv":
(Var("googlecode_url") % "libyuv") + "/trunk@389",
+ "third_party/opus/source":
+ "http://git.xiph.org/opus.git@v1.0.1",
+
"third_party/protobuf":
Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
diff --git a/src/build/common.gypi b/src/build/common.gypi
index 6d02f20..d57b265 100644
--- a/src/build/common.gypi
+++ b/src/build/common.gypi
@@ -85,6 +85,10 @@
# Disable the use of protocol buffers in production code.
'enable_protobuf%': 0,
+
+ # Disable Mozilla internal Opus version
+ 'build_with_mozilla%': 0,
+
}, { # Settings for the standalone (not-in-Chromium) build.
'include_pulse_audio%': 1,
'include_internal_audio_device%': 1,
diff --git a/src/engine_configurations.h b/src/engine_configurations.h
index 35dec15..553aed4 100644
--- a/src/engine_configurations.h
+++ b/src/engine_configurations.h
@@ -27,19 +27,27 @@
// [Voice] Codec settings
// ----------------------------------------------------------------------------
+// iSAC is not included in the Mozilla build, but in all other builds.
+#ifndef WEBRTC_MOZILLA_BUILD
#ifdef WEBRTC_ARCH_ARM
-#define WEBRTC_CODEC_ISACFX // fix-point iSAC implementation
+#define WEBRTC_CODEC_ISACFX // Fix-point iSAC implementation.
#else
-#define WEBRTC_CODEC_ISAC // floating-point iSAC implementation (default)
-#endif
+#define WEBRTC_CODEC_ISAC // Floating-point iSAC implementation (default).
+#endif // WEBRTC_ARCH_ARM
+#endif // !WEBRTC_MOZILLA_BUILD
+
+// AVT is included in all builds, along with G.711, NetEQ and CNG
+// (which are mandatory and don't have any defines).
#define WEBRTC_CODEC_AVT
-#ifndef WEBRTC_CHROMIUM_BUILD
+// iLBC, G.722, PCM16B and Redundancy coding are excluded from Chromium and
+// Mozilla builds.
+#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
#define WEBRTC_CODEC_ILBC
#define WEBRTC_CODEC_G722
#define WEBRTC_CODEC_PCM16
#define WEBRTC_CODEC_RED
-#endif
+#endif // !WEBRTC_CHROMIUM_BUILD && !WEBRTC_MOZILLA_BUILD
// ----------------------------------------------------------------------------
// [Video] Codec settings
diff --git a/src/modules/audio_coding/codecs/cng/webrtc_cng.c b/src/modules/audio_coding/codecs/cng/webrtc_cng.c
index 7cc6cb9..28bfaae 100644
--- a/src/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/src/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -36,7 +36,7 @@
typedef struct WebRtcCngEncInst_t_ {
int16_t enc_nrOfCoefs;
- int16_t enc_sampfreq;
+ uint16_t enc_sampfreq;
int16_t enc_interval;
int16_t enc_msSinceSID;
int32_t enc_Energy;
diff --git a/src/modules/audio_coding/codecs/opus/interface/opus_interface.h b/src/modules/audio_coding/codecs/opus/interface/opus_interface.h
new file mode 100644
index 0000000..dcfd87f
--- /dev/null
+++ b/src/modules/audio_coding/codecs/opus/interface/opus_interface.h
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+
+#include "typedefs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Opaque wrapper types for the codec state.
+typedef struct WebRtcOpusEncInst OpusEncInst;
+typedef struct WebRtcOpusDecInst OpusDecInst;
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels);
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Encode(...)
+ *
+ * This function encodes audio as a series of Opus frames and inserts
+ * it into a packet. Input buffer can be any length.
+ *
+ * Input:
+ * - inst : Encoder context
+ * - audio_in : Input speech data buffer
+ * - samples : Samples in audio_in
+ * - length_encoded_buffer : Output buffer size
+ *
+ * Output:
+ * - encoded : Output compressed data buffer
+ *
+ * Return value : >0 - Length (in bytes) of coded data
+ * -1 - Error
+ */
+int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
+ int16_t length_encoded_buffer, uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcOpus_SetBitRate(...)
+ *
+ * This function adjusts the target bitrate of the encoder.
+ *
+ * Input:
+ * - inst : Encoder context
+ * - rate : New target bitrate
+ *
+ * Return value : 0 - Success
+ * -1 - Error
+ */
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderInit(...)
+ *
+ * This function resets state of the decoder.
+ *
+ * Input:
+ * - inst : Decoder context
+ *
+ * Return value : 0 - Success
+ * -1 - Error
+ */
+int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Decode(...)
+ *
+ * This function decodes an Opus packet into one or more audio frames at the
+ * ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ * - inst : Decoder context
+ * - encoded : Encoded data
+ * - encoded_bytes : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - audio_type : 1 normal, 2 CNG (for Opus it should
+ * always return 1 since we're not using Opus's
+ * built-in DTX/CNG scheme)
+ *
+ * Return value : >0 - Samples in decoded vector
+ * -1 - Error
+ */
+int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
+ int16_t encoded_bytes, int16_t* decoded,
+ int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DecodePlc(...)
+ *
+ * This function precesses PLC for opus frame(s).
+ * Input:
+ * - inst : Decoder context
+ * - number_of_lost_frames : Number of PLC frames to produce
+ *
+ * Output:
+ * - decoded : The decoded vector
+ *
+ * Return value : >0 - number of samples in decoded PLC vector
+ * -1 - Error
+ */
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+ int16_t number_of_lost_frames);
+
+#ifdef __cplusplus
+} // extern "C"
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
diff --git a/src/modules/audio_coding/codecs/opus/opus.gypi b/src/modules/audio_coding/codecs/opus/opus.gypi
new file mode 100644
index 0000000..809068c
--- /dev/null
+++ b/src/modules/audio_coding/codecs/opus/opus.gypi
@@ -0,0 +1,44 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'webrtc_opus',
+ 'type': 'static_library',
+ 'conditions': [
+ ['build_with_mozilla==1', {
+ # Mozilla provides its own build of the opus library.
+ 'include_dirs': [
+ '$(DIST)/include/opus',
+ ]
+ }, {
+ 'dependencies': [
+ '<(DEPTH)/third_party/opus/opus.gyp:opus'
+ ],
+ 'include_dirs': [
+ '<(webrtc_root)/../third_party/opus/source/include',
+ ],
+ }],
+ ],
+ 'direct_dependent_settings': {
+ 'conditions': [
+ ['build_with_mozilla==1', {
+ 'include_dirs': [
+ '$(DIST)/include/opus',
+ ],
+ }],
+ ],
+ },
+ 'sources': [
+ 'interface/opus_interface.h',
+ 'opus_interface.c',
+ ],
+ },
+ ],
+}
diff --git a/src/modules/audio_coding/codecs/opus/opus_interface.c b/src/modules/audio_coding/codecs/opus/opus_interface.c
new file mode 100644
index 0000000..f61ecc5
--- /dev/null
+++ b/src/modules/audio_coding/codecs/opus/opus_interface.c
@@ -0,0 +1,181 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "opus.h"
+
+#include "common_audio/signal_processing/resample_by_2_internal.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+enum {
+ /* We always produce 20ms frames. */
+ kWebRtcOpusMaxEncodeFrameSizeMs = 20,
+
+ /* The format allows up to 120ms frames. Since we
+ * don't control the other side, we must allow
+ * for packets that large. NetEq is currently
+ * limited to 60 ms on the receive side.
+ */
+ kWebRtcOpusMaxDecodeFrameSizeMs = 120,
+
+ /* Sample count is 48 kHz * samples per frame. */
+ kWebRtcOpusMaxFrameSize = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
+};
+
+struct WebRtcOpusEncInst {
+ OpusEncoder* encoder;
+};
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
+ OpusEncInst* state;
+ state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
+ if (state) {
+ int error;
+ state->encoder = opus_encoder_create(48000, channels, OPUS_APPLICATION_VOIP,
+ &error);
+ if (error == OPUS_OK || state->encoder != NULL ) {
+ *inst = state;
+ return 0;
+ }
+ free(state);
+ }
+ return -1;
+}
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
+ opus_encoder_destroy(inst->encoder);
+ return 0;
+}
+
+int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
+ int16_t length_encoded_buffer, uint8_t* encoded) {
+ opus_int16* audio = (opus_int16*) audio_in;
+ unsigned char* coded = encoded;
+ int res;
+
+ if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
+ return -1;
+ }
+
+ res = opus_encode(inst->encoder, audio, samples, coded,
+ length_encoded_buffer);
+
+ if (res > 0) {
+ return res;
+ }
+ return -1;
+}
+
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
+}
+
+struct WebRtcOpusDecInst {
+ int16_t state_48_32[8];
+ OpusDecoder* decoder;
+};
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+ OpusDecInst* state;
+ state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
+ if (state) {
+ int error;
+ // Always create a 48000 Hz Opus decoder.
+ state->decoder = opus_decoder_create(48000, channels, &error);
+ if (error == OPUS_OK && state->decoder != NULL ) {
+ *inst = state;
+ return 0;
+ }
+ free(state);
+ state = NULL;
+ }
+ return -1;
+}
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
+ opus_decoder_destroy(inst->decoder);
+ free(inst);
+ return 0;
+}
+
+int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+ int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+ if (error == OPUS_OK) {
+ memset(inst->state_48_32, 0, sizeof(inst->state_48_32));
+ return 0;
+ }
+ return -1;
+}
+
+static int DecodeNative(OpusDecInst* inst, int16_t* encoded,
+ int16_t encoded_bytes, int16_t* decoded,
+ int16_t* audio_type) {
+ unsigned char* coded = (unsigned char*) encoded;
+ opus_int16* audio = (opus_int16*) decoded;
+
+ int res = opus_decode(inst->decoder, coded, encoded_bytes, audio,
+ kWebRtcOpusMaxFrameSize, 0);
+ /* TODO(tlegrand): set to DTX for zero-length packets? */
+ *audio_type = 0;
+
+ if (res > 0) {
+ return res;
+ }
+ return -1;
+}
+
+int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
+ int16_t encoded_bytes, int16_t* decoded,
+ int16_t* audio_type) {
+ /* Enough for 120 ms (the largest Opus packet size) of mono audio at 48 kHz
+ * and resampler overlap. This will need to be enlarged for stereo decoding.
+ */
+ int16_t buffer16[kWebRtcOpusMaxFrameSize];
+ int32_t buffer32[kWebRtcOpusMaxFrameSize + 7];
+ int decoded_samples;
+ int blocks;
+ int16_t output_samples;
+ int i;
+
+ /* Decode to a temporary buffer. */
+ decoded_samples = DecodeNative(inst, encoded, encoded_bytes, buffer16,
+ audio_type);
+ if (decoded_samples < 0) {
+ return -1;
+ }
+ /* Resample from 48 kHz to 32 kHz. */
+ for (i = 0; i < 7; i++) {
+ buffer32[i] = inst->state_48_32[i];
+ inst->state_48_32[i] = buffer16[decoded_samples -7 + i];
+ }
+ for (i = 0; i < decoded_samples; i++) {
+ buffer32[7 + i] = buffer16[i];
+ }
+ /* Resampling 3 samples to 2. Function divides the input in |blocks| number
+ * of 3-sample groups, and output is |blocks| number of 2-sample groups. */
+ blocks = decoded_samples / 3;
+ WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, blocks);
+ output_samples = (int16_t) (blocks * 2);
+ WebRtcSpl_VectorBitShiftW32ToW16(decoded, output_samples, buffer32, 15);
+
+ return output_samples;
+}
+
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+ int16_t number_of_lost_frames) {
+ /* TODO(tlegrand): We can pass NULL to opus_decode to activate packet
+ * loss concealment, but I don't know how many samples
+ * number_of_lost_frames corresponds to. */
+ return -1;
+}
diff --git a/src/modules/audio_coding/main/source/acm_cng.cc b/src/modules/audio_coding/main/source/acm_cng.cc
index 2393346..4edfc05 100644
--- a/src/modules/audio_coding/main/source/acm_cng.cc
+++ b/src/modules/audio_coding/main/source/acm_cng.cc
@@ -81,7 +81,8 @@
// Then return the structure back to NetEQ to add the codec to it's
// database.
- if (_sampFreqHz == 8000 || _sampFreqHz == 16000 || _sampFreqHz == 32000) {
+ if (_sampFreqHz == 8000 || _sampFreqHz == 16000 || _sampFreqHz == 32000 ||
+ _sampFreqHz == 48000) {
SET_CODEC_PAR((codecDef), kDecoderCNG, codecInst.pltype,
_decoderInstPtr, _sampFreqHz);
SET_CNG_FUNCTIONS((codecDef));
diff --git a/src/modules/audio_coding/main/source/acm_cng.h b/src/modules/audio_coding/main/source/acm_cng.h
index 6276c44..d204d02 100644
--- a/src/modules/audio_coding/main/source/acm_cng.h
+++ b/src/modules/audio_coding/main/source/acm_cng.h
@@ -62,7 +62,7 @@
WebRtcCngEncInst* _encoderInstPtr;
WebRtcCngDecInst* _decoderInstPtr;
- WebRtc_Word16 _sampFreqHz;
+ WebRtc_UWord16 _sampFreqHz;
};
} // namespace webrtc
diff --git a/src/modules/audio_coding/main/source/acm_codec_database.cc b/src/modules/audio_coding/main/source/acm_codec_database.cc
index b782194..2e3db19 100644
--- a/src/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/src/modules/audio_coding/main/source/acm_codec_database.cc
@@ -86,6 +86,10 @@
#include "acm_gsmfr.h"
#include "gsmfr_interface.h"
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ #include "acm_opus.h"
+ #include "opus_interface.h"
+#endif
#ifdef WEBRTC_CODEC_SPEEX
#include "acm_speex.h"
#include "speex_interface.h"
@@ -103,22 +107,20 @@
// codecs. Note! There are a limited number of payload types. If more codecs
// are defined they will receive reserved fixed payload types (values 69-95).
const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
- 105, 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 120,
- 121, 122, 123, 124, 125, 126, 101, 100, 97, 96, 95, 94,
- 93, 92, 91, 90, 89, 88, 87, 86, 85, 84, 83, 82,
- 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70,
- 69,
+ 105, 107, 108, 109, 111, 112, 113, 114, 115, 116, 117, 121,
+ 92, 91, 90, 89, 88, 87, 86, 85, 84, 83, 82, 81,
+ 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69,
+ 68, 67
};
// Creates database with all supported codecs at compile time.
// Each entry needs the following parameters in the given order:
// payload type, name, sampling frequency, packet size in samples,
// number of channels, and default rate.
-#if (defined(WEBRTC_CODEC_PCM16) || \
- defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) || \
- defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G729_1) || \
- defined(WEBRTC_CODEC_SPEEX) || defined(WEBRTC_CODEC_G722_1) || \
- defined(WEBRTC_CODEC_G722_1C))
+#if (defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) \
+ || defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G722_1) \
+ || defined(WEBRTC_CODEC_G722_1C) || defined(WEBRTC_CODEC_G729_1) \
+ || defined(WEBRTC_CODEC_PCM16) || defined(WEBRTC_CODEC_SPEEX))
static int count_database = 0;
#endif
@@ -186,14 +188,19 @@
#ifdef WEBRTC_CODEC_GSMFR
{3, "GSM", 8000, 160, 1, 13200},
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ // Opus supports 48, 24, 16, 12, 8 kHz.
+ {120, "opus", 48000, 960, 1, 32000},
+#endif
#ifdef WEBRTC_CODEC_SPEEX
{kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
{kDynamicPayloadtypes[count_database++], "speex", 16000, 320, 1, 22000},
#endif
- // Comfort noise for three different sampling frequencies.
+ // Comfort noise for four different sampling frequencies.
{13, "CN", 8000, 240, 1, 0},
{98, "CN", 16000, 480, 1, 0},
{99, "CN", 32000, 960, 1, 0},
+ {100, "CN", 48000, 1440, 1, 0},
#ifdef WEBRTC_CODEC_AVT
{106, "telephone-event", 8000, 240, 1, 0},
#endif
@@ -272,6 +279,11 @@
#ifdef WEBRTC_CODEC_GSMFR
{3, {160, 320, 480}, 160, 1},
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ // Opus supports frames shorter than 10ms,
+ // but it doesn't help us to use them.
+ {1, {960}, 0, 2},
+#endif
#ifdef WEBRTC_CODEC_SPEEX
{3, {160, 320, 480}, 0, 1},
{3, {320, 640, 960}, 0, 1},
@@ -280,6 +292,7 @@
{1, {240}, 240, 1},
{1, {480}, 480, 1},
{1, {960}, 960, 1},
+ {1, {1440}, 1440, 1},
#ifdef WEBRTC_CODEC_AVT
{1, {240}, 240, 1},
#endif
@@ -355,6 +368,9 @@
#ifdef WEBRTC_CODEC_GSMFR
kDecoderGSMFR,
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ kDecoderOpus,
+#endif
#ifdef WEBRTC_CODEC_SPEEX
kDecoderSPEEX_8,
kDecoderSPEEX_16,
@@ -363,6 +379,7 @@
kDecoderCNG,
kDecoderCNG,
kDecoderCNG,
+ kDecoderCNG,
#ifdef WEBRTC_CODEC_AVT
kDecoderAVT,
#endif
@@ -509,6 +526,9 @@
} else if (STR_CASE_CMP("g7291", codec_inst->plname) == 0) {
return IsG7291RateValid(codec_inst->rate)
? codec_id : kInvalidRate;
+ } else if (STR_CASE_CMP("opus", codec_inst->plname) == 0) {
+ return IsOpusRateValid(codec_inst->rate)
+ ? codec_id : kInvalidRate;
} else if (STR_CASE_CMP("speex", codec_inst->plname) == 0) {
return IsSpeexRateValid(codec_inst->rate)
? codec_id : kInvalidRate;
@@ -719,6 +739,10 @@
codec_id = kCNSWB;
break;
}
+ case 48000: {
+ codec_id = kCNFB;
+ break;
+ }
default: {
return NULL;
}
@@ -732,6 +756,10 @@
#ifdef WEBRTC_CODEC_G729_1
return new ACMG729_1(kG729_1);
#endif
+ } else if (!STR_CASE_CMP(codec_inst->plname, "opus")) {
+#ifdef WEBRTC_CODEC_OPUS
+ return new ACMOpus(kOpus);
+#endif
} else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
#ifdef WEBRTC_CODEC_SPEEX
int codec_id;
@@ -766,6 +794,10 @@
codec_id = kCNSWB;
break;
}
+ case 48000: {
+ codec_id = kCNFB;
+ break;
+ }
default: {
return NULL;
}
@@ -928,6 +960,14 @@
}
}
+// Checks if the bitrate is valid for Opus.
+bool ACMCodecDB::IsOpusRateValid(int rate) {
+ if ((rate < 6000) || (rate > 510000)) {
+ return false;
+ }
+ return true;
+}
+
// Checks if the bitrate is valid for Celt.
bool ACMCodecDB::IsCeltRateValid(int rate) {
if ((rate >= 48000) && (rate <= 128000)) {
diff --git a/src/modules/audio_coding/main/source/acm_codec_database.h b/src/modules/audio_coding/main/source/acm_codec_database.h
index 0fe3a5e..8baf24e 100644
--- a/src/modules/audio_coding/main/source/acm_codec_database.h
+++ b/src/modules/audio_coding/main/source/acm_codec_database.h
@@ -91,6 +91,9 @@
#ifdef WEBRTC_CODEC_GSMFR
, kGSMFR
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ , kOpus
+#endif
#ifdef WEBRTC_CODEC_SPEEX
, kSPEEX8
, kSPEEX16
@@ -98,6 +101,7 @@
, kCNNB
, kCNWB
, kCNSWB
+ , kCNFB
#ifdef WEBRTC_CODEC_AVT
, kAVT
#endif
@@ -170,6 +174,9 @@
enum {kSPEEX8 = -1};
enum {kSPEEX16 = -1};
#endif
+#ifndef WEBRTC_CODEC_OPUS
+ enum {kOpus = -1};
+#endif
#ifndef WEBRTC_CODEC_AVT
enum {kAVT = -1};
#endif
@@ -298,6 +305,7 @@
static bool IsAMRwbRateValid(int rate);
static bool IsG7291RateValid(int rate);
static bool IsSpeexRateValid(int rate);
+ static bool IsOpusRateValid(int rate);
static bool IsCeltRateValid(int rate);
// Check if the payload type is valid, meaning that it is in the valid range
diff --git a/src/modules/audio_coding/main/source/acm_common_defs.h b/src/modules/audio_coding/main/source/acm_common_defs.h
index fd8dbd6..cdff1c1 100644
--- a/src/modules/audio_coding/main/source/acm_common_defs.h
+++ b/src/modules/audio_coding/main/source/acm_common_defs.h
@@ -63,14 +63,15 @@
// kPassiveDTXNB : Passive audio frame coded by narrow-band CN.
// kPassiveDTXWB : Passive audio frame coded by wide-band CN.
// kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN.
-//
+// kPassiveDTXFB : Passive audio frame coded by full-band CN.
enum WebRtcACMEncodingType {
kNoEncoding,
kActiveNormalEncoded,
kPassiveNormalEncoded,
kPassiveDTXNB,
kPassiveDTXWB,
- kPassiveDTXSWB
+ kPassiveDTXSWB,
+ kPassiveDTXFB
};
// A structure which contains codec parameters. For instance, used when
diff --git a/src/modules/audio_coding/main/source/acm_generic_codec.cc b/src/modules/audio_coding/main/source/acm_generic_codec.cc
index f9a6a3a..f98f260 100644
--- a/src/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/src/modules/audio_coding/main/source/acm_generic_codec.cc
@@ -58,6 +58,7 @@
_numLPCParams(kNewCNGNumPLCParams),
_sentCNPrevious(false),
_isMaster(true),
+ _prev_frame_cng(0),
_netEqDecodeLock(NULL),
_codecWrapperLock(*RWLockWrapper::CreateRWLock()),
_lastEncodedTimestamp(0),
@@ -294,6 +295,8 @@
*encodingType = kPassiveDTXWB;
} else if (sampFreqHz == 32000) {
*encodingType = kPassiveDTXSWB;
+ } else if (sampFreqHz == 48000) {
+ *encodingType = kPassiveDTXFB;
} else {
status = -1;
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
@@ -1169,7 +1172,7 @@
}
WebRtc_UWord16 freqHz;
EncoderSampFreq(freqHz);
- if(WebRtcCng_InitEnc(_ptrDTXInst, (WebRtc_Word16)freqHz,
+ if(WebRtcCng_InitEnc(_ptrDTXInst, freqHz,
ACM_SID_INTERVAL_MSEC, _numLPCParams) < 0)
{
// Couldn't initialize, has to return -1, and free the memory
@@ -1313,6 +1316,7 @@
*samplesProcessed = 0;
return 0;
}
+
WebRtc_UWord16 freqHz;
EncoderSampFreq(freqHz);
@@ -1321,8 +1325,8 @@
WebRtc_Word32 frameLenMsec = (((WebRtc_Word32)_frameLenSmpl * 1000) / freqHz);
WebRtc_Word16 status;
- // Vector for storing maximum 30 ms of mono audio at 32 kHz
- WebRtc_Word16 audio[960];
+ // Vector for storing maximum 30 ms of mono audio at 48 kHz.
+ WebRtc_Word16 audio[1440];
// Calculate number of VAD-blocks to process, and number of samples in each block.
int noSamplesToProcess[2];
@@ -1378,25 +1382,33 @@
*bitStreamLenByte = 0;
for(WebRtc_Word16 n = 0; n < num10MsecFrames; n++)
{
- // This block is (passive) && (vad enabled)
- status = WebRtcCng_Encode(_ptrDTXInst, &audio[n*samplesIn10Msec],
- samplesIn10Msec, bitStream, &bitStreamLen, 0);
+ // This block is (passive) && (vad enabled). If first CNG after
+ // speech, force SID by setting last parameter to "1".
+ status = WebRtcCng_Encode(_ptrDTXInst,
+ &audio[n*samplesIn10Msec],
+ samplesIn10Msec, bitStream,
+ &bitStreamLen, !_prev_frame_cng);
if (status < 0) {
return -1;
}
+ // Update previous frame was CNG.
+ _prev_frame_cng = 1;
+
*samplesProcessed += samplesIn10Msec*_noChannels;
// bitStreamLen will only be > 0 once per 100 ms
*bitStreamLenByte += bitStreamLen;
}
-
// Check if all samples got processed by the DTX
if(*samplesProcessed != noSamplesToProcess[i]*_noChannels) {
// Set to zero since something went wrong. Shouldn't happen.
*samplesProcessed = 0;
}
+ } else {
+ // Update previous frame was not CNG.
+ _prev_frame_cng = 0;
}
if(*samplesProcessed > 0)
diff --git a/src/modules/audio_coding/main/source/acm_generic_codec.h b/src/modules/audio_coding/main/source/acm_generic_codec.h
index c138ed9..29c882c 100644
--- a/src/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/src/modules/audio_coding/main/source/acm_generic_codec.h
@@ -1310,6 +1310,7 @@
WebRtc_UWord8 _numLPCParams;
bool _sentCNPrevious;
bool _isMaster;
+ int16_t _prev_frame_cng;
WebRtcACMCodecParams _encoderParams;
WebRtcACMCodecParams _decoderParams;
diff --git a/src/modules/audio_coding/main/source/acm_opus.cc b/src/modules/audio_coding/main/source/acm_opus.cc
index 87bdd8b..034e57d 100644
--- a/src/modules/audio_coding/main/source/acm_opus.cc
+++ b/src/modules/audio_coding/main/source/acm_opus.cc
@@ -8,442 +8,256 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "acm_opus.h"
+
+#include "acm_codec_database.h"
#include "acm_common_defs.h"
#include "acm_neteq.h"
-#include "acm_opus.h"
#include "trace.h"
#include "webrtc_neteq.h"
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_OPUS
- // NOTE! Opus is not included in the open-source package. Modify this file or your codec
- // API to match the function call and name of used Opus API file.
- // #include "opus_interface.h"
+#include "opus_interface.h"
#endif
-namespace webrtc
-{
+namespace webrtc {
#ifndef WEBRTC_CODEC_OPUS
-ACMOPUS::ACMOPUS(WebRtc_Word16 /* codecID */)
+ACMOpus::ACMOpus(int16_t /* codecID */)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL),
- _mySampFreq(0),
- _myRate(0),
- _opusMode(0),
- _flagVBR(0) {
+ _sampleFreq(0),
+ _bitrate(0) {
return;
}
-
-ACMOPUS::~ACMOPUS()
-{
- return;
+ACMOpus::~ACMOpus() {
+ return;
}
-
-WebRtc_Word16
-ACMOPUS::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
+int16_t ACMOpus::InternalEncode(uint8_t* /* bitStream */,
+ int16_t* /* bitStreamLenByte */) {
+ return -1;
}
-
-WebRtc_Word16
-ACMOPUS::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
+int16_t ACMOpus::DecodeSafe(uint8_t* /* bitStream */,
+ int16_t /* bitStreamLenByte */,
+ int16_t* /* audio */,
+ int16_t* /* audioSamples */,
+ int8_t* /* speechType */) {
+ return -1;
}
-
-WebRtc_Word16
-ACMOPUS::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
+int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
}
-
-WebRtc_Word16
-ACMOPUS::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
+int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* /* codecParams */) {
+ return -1;
}
-
-WebRtc_Word32
-ACMOPUS::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
+int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
+ const CodecInst& /* codecInst */) {
+ return -1;
}
-
-ACMGenericCodec*
-ACMOPUS::CreateInstance(void)
-{
- return NULL;
+ACMGenericCodec* ACMOpus::CreateInstance(void) {
+ return NULL;
}
-
-WebRtc_Word16
-ACMOPUS::InternalCreateEncoder()
-{
- return -1;
+int16_t ACMOpus::InternalCreateEncoder() {
+ return -1;
}
-
-void
-ACMOPUS::DestructEncoderSafe()
-{
- return;
+void ACMOpus::DestructEncoderSafe() {
+ return;
}
-
-WebRtc_Word16
-ACMOPUS::InternalCreateDecoder()
-{
- return -1;
+int16_t ACMOpus::InternalCreateDecoder() {
+ return -1;
}
-
-void
-ACMOPUS::DestructDecoderSafe()
-{
- return;
+void ACMOpus::DestructDecoderSafe() {
+ return;
}
-
-void
-ACMOPUS::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
+void ACMOpus::InternalDestructEncoderInst(void* /* ptrInst */) {
+ return;
}
-WebRtc_Word16
-ACMOPUS::SetBitRateSafe(
- const WebRtc_Word32 /*rate*/ )
-{
- return -1;
+int16_t ACMOpus::SetBitRateSafe(const int32_t /*rate*/) {
+ return -1;
}
-#else //===================== Actual Implementation =======================
+#else //===================== Actual Implementation =======================
-// Remove when integrating a real Opus wrapper
-extern WebRtc_Word16 WebRtcOpus_CreateEnc(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
-extern WebRtc_Word16 WebRtcOpus_CreateDec(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
-extern WebRtc_Word16 WebRtcOpus_FreeEnc(OPUS_inst_t_* inst);
-extern WebRtc_Word16 WebRtcOpus_FreeDec(OPUS_inst_t_* inst);
-extern WebRtc_Word16 WebRtcOpus_Encode(OPUS_inst_t_* encInst,
- WebRtc_Word16* input,
- WebRtc_Word16* output,
- WebRtc_Word16 len,
- WebRtc_Word16 byteLen);
-extern WebRtc_Word16 WebRtcOpus_EncoderInit(OPUS_inst_t_* encInst,
- WebRtc_Word16 samplFreq,
- WebRtc_Word16 mode,
- WebRtc_Word16 vbrFlag);
-extern WebRtc_Word16 WebRtcOpus_Decode(OPUS_inst_t_* decInst);
-extern WebRtc_Word16 WebRtcOpus_DecodeBwe(OPUS_inst_t_* decInst, WebRtc_Word16* input);
-extern WebRtc_Word16 WebRtcOpus_DecodePlc(OPUS_inst_t_* decInst);
-extern WebRtc_Word16 WebRtcOpus_DecoderInit(OPUS_inst_t_* decInst);
-
-ACMOPUS::ACMOPUS(WebRtc_Word16 codecID)
+ACMOpus::ACMOpus(int16_t codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL),
- _mySampFreq(48000), // Default sampling frequency.
- _myRate(50000), // Default rate.
- _opusMode(1), // Default mode is the hybrid mode.
- _flagVBR(0) { // Default VBR off.
+ _sampleFreq(32000), // Default sampling frequency.
+ _bitrate(20000) { // Default bit-rate.
_codecID = codecID;
-
- // Current implementation doesn't have DTX. That might change.
+ // Opus has internal DTX, but we dont use it for now.
_hasInternalDTX = false;
+ if (_codecID != ACMCodecDB::kOpus) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Wrong codec id for Opus.");
+ _sampleFreq = -1;
+ _bitrate = -1;
+ }
return;
}
-ACMOPUS::~ACMOPUS()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcOpus_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcOpus_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
+ACMOpus::~ACMOpus() {
+ if (_encoderInstPtr != NULL) {
+ WebRtcOpus_EncoderFree(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ if (_decoderInstPtr != NULL) {
+ WebRtcOpus_DecoderFree(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ return;
}
-
-WebRtc_Word16
-ACMOPUS::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- WebRtc_Word16 noEncodedSamples = 0;
- WebRtc_Word16 tmpLenByte = 0;
+int16_t ACMOpus::InternalEncode(uint8_t* bitStream, int16_t* bitStreamLenByte) {
+ // Call Encoder.
+ *bitStreamLenByte = WebRtcOpus_Encode(_encoderInstPtr,
+ &_inAudio[_inAudioIxRead],
+ _frameLenSmpl,
+ MAX_PAYLOAD_SIZE_BYTE,
+ bitStream);
+ // Check for error reported from encoder.
+ if (*bitStreamLenByte < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "InternalEncode: Encode error for Opus");
*bitStreamLenByte = 0;
+ return -1;
+ }
- WebRtc_Word16 byteLengthFrame = 0;
+ // Increment the read index. This tells the caller how far
+ // we have gone forward in reading the audio buffer.
+ _inAudioIxRead += _frameLenSmpl;
- // Derive what byte-length is requested
- byteLengthFrame = _myRate*_frameLenSmpl/(8*_mySampFreq);
-
- // Call Encoder
- *bitStreamLenByte = WebRtcOpus_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
- (WebRtc_Word16*)bitStream, _frameLenSmpl, byteLengthFrame);
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl;
-
- // sanity check
- if(*bitStreamLenByte < 0)
- {
- // error has happened
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Encode error for Opus");
- *bitStreamLenByte = 0;
- return -1;
- }
-
- return *bitStreamLenByte;
+ return *bitStreamLenByte;
}
+int16_t ACMOpus::DecodeSafe(uint8_t* bitStream, int16_t bitStreamLenByte,
+ int16_t* audio, int16_t* audioSamples,
+ int8_t* speechType) {
+ return 0;
+}
+int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+ int16_t ret;
+ if (_encoderInstPtr != NULL) {
+ WebRtcOpus_EncoderFree(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+ ret = WebRtcOpus_EncoderCreate(&_encoderInstPtr,
+ codecParams->codecInstant.channels);
+ if (ret < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Encoder creation failed for Opus");
+ return ret;
+ }
+ ret = WebRtcOpus_SetBitRate(_encoderInstPtr, codecParams->codecInstant.rate);
+ if (ret < 0) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "Setting initial bitrate failed for Opus");
+ return ret;
+ }
+ return 0;
+}
-WebRtc_Word16
-ACMOPUS::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
+int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
+ if (_decoderInstPtr != NULL) {
+ WebRtcOpus_DecoderFree(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+ if (WebRtcOpus_DecoderCreate(&_decoderInstPtr,
+ codecParams->codecInstant.channels) < 0) {
+ return -1;
+ }
+ return WebRtcOpus_DecoderInit(_decoderInstPtr);
+}
+
+int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+ const CodecInst& codecInst) {
+ if (!_decoderInitialized) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "CodeDef: Decoder uninitialized for Opus");
+ return -1;
+ }
+
+ // Fill up the structure by calling "SET_CODEC_PAR" & "SET_OPUS_FUNCTION."
+ // Then call NetEQ to add the codec to its database.
+ // TODO(tlegrand): Decoder is registered in NetEQ as a 32 kHz decoder, which
+ // is true until we have a full 48 kHz system, and remove the downsampling
+ // in the Opus decoder wrapper.
+ SET_CODEC_PAR((codecDef), kDecoderOpus, codecInst.pltype, _decoderInstPtr,
+ 32000);
+ SET_OPUS_FUNCTIONS((codecDef));
+ return 0;
+}
+
+ACMGenericCodec* ACMOpus::CreateInstance(void) {
+ return NULL;
+}
+
+int16_t ACMOpus::InternalCreateEncoder() {
+ // Real encoder will be created in InternalInitEncoder.
+ return 0;
+}
+
+void ACMOpus::DestructEncoderSafe() {
+ if (_encoderInstPtr) {
+ WebRtcOpus_EncoderFree(_encoderInstPtr);
+ _encoderInstPtr = NULL;
+ }
+}
+
+int16_t ACMOpus::InternalCreateDecoder() {
+ // Real decoder will be created in InternalInitDecoder
+ return 0;
+}
+
+void ACMOpus::DestructDecoderSafe() {
+ _decoderInitialized = false;
+ if (_decoderInstPtr) {
+ WebRtcOpus_DecoderFree(_decoderInstPtr);
+ _decoderInstPtr = NULL;
+ }
+}
+
+void ACMOpus::InternalDestructEncoderInst(void* ptrInst) {
+ if (ptrInst != NULL) {
+ WebRtcOpus_EncoderFree((OpusEncInst*) ptrInst);
+ }
+ return;
+}
+
+int16_t ACMOpus::SetBitRateSafe(const int32_t rate) {
+ if (rate < 6000 || rate > 510000) {
+ WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+ "SetBitRateSafe: Invalid rate Opus");
+ return -1;
+ }
+
+ _bitrate = rate;
+
+ // Ask the encoder for the new rate.
+ if (WebRtcOpus_SetBitRate(_encoderInstPtr, _bitrate) >= 0) {
+ _encoderParams.codecInstant.rate = _bitrate;
return 0;
+ }
+
+ return -1;
}
+#endif // WEBRTC_CODEC_OPUS
-WebRtc_Word16
-ACMOPUS::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- //set the bit rate and initialize
- _myRate = codecParams->codecInstant.rate;
- return SetBitRateSafe( (WebRtc_UWord32)_myRate);
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- if (WebRtcOpus_DecoderInit(_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: init decoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMOPUS::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: Decoder uninitialized for Opus");
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderOpus, codecInst.pltype,
- _decoderInstPtr, 16000);
- SET_OPUS_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMOPUS::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateEncoder()
-{
- if (WebRtcOpus_CreateEnc(&_encoderInstPtr, _mySampFreq) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: create encoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMOPUS::DestructEncoderSafe()
-{
- _encoderExist = false;
- _encoderInitialized = false;
- if(_encoderInstPtr != NULL)
- {
- WebRtcOpus_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateDecoder()
-{
- if (WebRtcOpus_CreateDec(&_decoderInstPtr, _mySampFreq) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: create decoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMOPUS::DestructDecoderSafe()
-{
- _decoderExist = false;
- _decoderInitialized = false;
- if(_decoderInstPtr != NULL)
- {
- WebRtcOpus_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
-}
-
-
-void
-ACMOPUS::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcOpus_FreeEnc((OPUS_inst_t*)ptrInst);
- }
- return;
-}
-
-WebRtc_Word16
-ACMOPUS::SetBitRateSafe(
- const WebRtc_Word32 rate)
-{
- //allowed rates: {8000, 12000, 14000, 16000, 18000, 20000,
- // 22000, 24000, 26000, 28000, 30000, 32000};
- switch(rate)
- {
- case 8000:
- {
- _myRate = 8000;
- break;
- }
- case 12000:
- {
- _myRate = 12000;
- break;
- }
- case 14000:
- {
- _myRate = 14000;
- break;
- }
- case 16000:
- {
- _myRate = 16000;
- break;
- }
- case 18000:
- {
- _myRate = 18000;
- break;
- }
- case 20000:
- {
- _myRate = 20000;
- break;
- }
- case 22000:
- {
- _myRate = 22000;
- break;
- }
- case 24000:
- {
- _myRate = 24000;
- break;
- }
- case 26000:
- {
- _myRate = 26000;
- break;
- }
- case 28000:
- {
- _myRate = 28000;
- break;
- }
- case 30000:
- {
- _myRate = 30000;
- break;
- }
- case 32000:
- {
- _myRate = 32000;
- break;
- }
- default:
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: Invalid rate Opus");
- return -1;
- }
- }
-
- // Re-init with new rate
- if (WebRtcOpus_EncoderInit(_encoderInstPtr, _mySampFreq, _opusMode, _flagVBR) >= 0)
- {
- _encoderParams.codecInstant.rate = _myRate;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-#endif
-
-} // namespace webrtc
+} // namespace webrtc
diff --git a/src/modules/audio_coding/main/source/acm_opus.h b/src/modules/audio_coding/main/source/acm_opus.h
index c6832fa..d8baa30 100644
--- a/src/modules/audio_coding/main/source/acm_opus.h
+++ b/src/modules/audio_coding/main/source/acm_opus.h
@@ -12,68 +12,48 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
#include "acm_generic_codec.h"
+#include "opus_interface.h"
+#include "resampler.h"
-// forward declaration
-struct OPUS_inst_t_;
-struct OPUS_inst_t_;
+namespace webrtc {
-namespace webrtc
-{
+class ACMOpus : public ACMGenericCodec {
+ public:
+ ACMOpus(int16_t codecID);
+ ~ACMOpus();
-class ACMOPUS: public ACMGenericCodec
-{
-public:
- ACMOPUS(WebRtc_Word16 codecID);
- ~ACMOPUS();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
+ ACMGenericCodec* CreateInstance(void);
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
+ int16_t InternalEncode(uint8_t* bitstream, int16_t* bitStreamLenByte);
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
+ int16_t InternalInitEncoder(WebRtcACMCodecParams *codecParams);
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
+ int16_t InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
+ protected:
+ int16_t DecodeSafe(uint8_t* bitStream, int16_t bitStreamLenByte,
+ int16_t* audio, int16_t* audioSamples, int8_t* speechType);
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
+ int32_t CodecDef(WebRtcNetEQ_CodecDef& codecDef, const CodecInst& codecInst);
- void DestructEncoderSafe();
+ void DestructEncoderSafe();
- void DestructDecoderSafe();
+ void DestructDecoderSafe();
- WebRtc_Word16 InternalCreateEncoder();
+ int16_t InternalCreateEncoder();
- WebRtc_Word16 InternalCreateDecoder();
+ int16_t InternalCreateDecoder();
- void InternalDestructEncoderInst(
- void* ptrInst);
+ void InternalDestructEncoderInst(void* ptrInst);
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 rate);
+ int16_t SetBitRateSafe(const int32_t rate);
- OPUS_inst_t_* _encoderInstPtr;
- OPUS_inst_t_* _decoderInstPtr;
-
- WebRtc_UWord16 _mySampFreq;
- WebRtc_UWord16 _myRate;
- WebRtc_Word16 _opusMode;
- WebRtc_Word16 _flagVBR;
-
+ OpusEncInst* _encoderInstPtr;
+ OpusDecInst* _decoderInstPtr;
+ uint16_t _sampleFreq;
+ uint16_t _bitrate;
};
-} // namespace webrtc
+} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
diff --git a/src/modules/audio_coding/main/source/audio_coding_module.gypi b/src/modules/audio_coding/main/source/audio_coding_module.gypi
index bc9ea7d..f62ba36 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module.gypi
+++ b/src/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -15,6 +15,7 @@
'iLBC',
'iSAC',
'iSACFix',
+ 'webrtc_opus',
'PCM16B',
'NetEq',
'<(webrtc_root)/common_audio/common_audio.gyp:resampler',
@@ -37,11 +38,13 @@
'include_dirs': [
'../interface',
'../../../interface',
+ '../../codecs/opus/interface',
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
'../../../interface',
+ '../../codecs/opus/interface',
],
},
'sources': [
diff --git a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
index c1341b9..0a399ba 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -45,6 +45,7 @@
_cng_nb_pltype(255),
_cng_wb_pltype(255),
_cng_swb_pltype(255),
+ _cng_fb_pltype(255),
_red_pltype(255),
_vadEnabled(false),
_dtxEnabled(false),
@@ -112,6 +113,8 @@
_cng_wb_pltype = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
} else if (ACMCodecDB::database_[i].plfreq == 32000) {
_cng_swb_pltype = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
+ } else if (ACMCodecDB::database_[i].plfreq == 48000) {
+ _cng_fb_pltype = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
}
}
}
@@ -320,6 +323,12 @@
_isFirstRED = true;
break;
}
+ case kPassiveDTXFB: {
+ current_payload_type = _cng_fb_pltype;
+ frame_type = kAudioFrameCN;
+ _isFirstRED = true;
+ break;
+ }
}
has_data_to_send = true;
_previousPayloadType = current_payload_type;
@@ -612,6 +621,10 @@
_cng_swb_pltype = static_cast<uint8_t>(send_codec.pltype);
break;
}
+ case 48000: {
+ _cng_fb_pltype = static_cast<uint8_t>(send_codec.pltype);
+ break;
+ }
default: {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
"RegisterSendCodec() failed, invalid frequency for CNG "
@@ -1254,6 +1267,9 @@
CriticalSectionScoped lock(_acmCritSect);
if (DecoderParamByPlType(_lastRecvAudioCodecPlType, codec_params) < 0) {
return _netEq.CurrentSampFreqHz();
+ } else if (codec_params.codecInstant.plfreq == 48000) {
+ // TODO(tlegrand): Remove this option when we have full 48 kHz support.
+ return 32000;
} else {
return codec_params.codecInstant.plfreq;
}
diff --git a/src/modules/audio_coding/main/source/audio_coding_module_impl.h b/src/modules/audio_coding/main/source/audio_coding_module_impl.h
index 145faf6..0e7f2f3 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/src/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -279,6 +279,7 @@
uint8_t _cng_nb_pltype;
uint8_t _cng_wb_pltype;
uint8_t _cng_swb_pltype;
+ uint8_t _cng_fb_pltype;
uint8_t _red_pltype;
bool _vadEnabled;
bool _dtxEnabled;
diff --git a/src/modules/audio_coding/main/test/TestAllCodecs.cc b/src/modules/audio_coding/main/test/TestAllCodecs.cc
index b312390..89a9829 100644
--- a/src/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/src/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -614,6 +614,28 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_opus[] = "OPUS";
+ RegisterSendCodec('A', codec_opus, 48000, 6000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 20000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 32000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 48000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 64000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 96000, 960, -1);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 500000, 960, -1);
+ Run(channel_a_to_b_);
+#endif
if (test_mode_ != 0) {
printf("===============================================================\n");
diff --git a/src/modules/audio_coding/main/test/TestVADDTX.cc b/src/modules/audio_coding/main/test/TestVADDTX.cc
index 793ab579..fcca374 100644
--- a/src/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/src/modules/audio_coding/main/test/TestVADDTX.cc
@@ -138,6 +138,21 @@
_outFileB.Close();
#endif
+#ifdef WEBRTC_CODEC_OPUS
+ // Open outputfile
+ OpenOutFile(testCntr++);
+
+ // Register Opus as send codec
+ char nameOPUS[] = "opus";
+ RegisterSendCodec('A', nameOPUS);
+
+ // Run the five test cased
+ runTestCases();
+
+ // Close file
+ _outFileB.Close();
+
+#endif
if(_testMode) {
printf("Done!\n");
}
diff --git a/src/modules/audio_coding/neteq/codec_db.c b/src/modules/audio_coding/neteq/codec_db.c
index 5369cfd..ebc9216 100644
--- a/src/modules/audio_coding/neteq/codec_db.c
+++ b/src/modules/audio_coding/neteq/codec_db.c
@@ -84,7 +84,7 @@
#ifdef NETEQ_32KHZ_WIDEBAND
&&(codec_fs!=32000)
#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
+#if defined(NETEQ_48KHZ_WIDEBAND) || defined(NETEQ_OPUS_CODEC)
&&(codec_fs!=48000)
#endif
)
@@ -114,6 +114,9 @@
#ifdef NETEQ_ISAC_SWB_CODEC
case kDecoderISACswb :
#endif
+#ifdef NETEQ_OPUS_CODEC
+ case kDecoderOpus :
+#endif
#ifdef NETEQ_G722_CODEC
case kDecoderG722 :
case kDecoderG722_2ch :
@@ -458,6 +461,9 @@
#ifdef NETEQ_ISAC_SWB_CODEC
case kDecoderISACswb:
#endif
+#ifdef NETEQ_OPUS_CODEC
+ case kDecoderOpus:
+#endif
#ifdef NETEQ_ARBITRARY_CODEC
case kDecoderArbitrary:
#endif
diff --git a/src/modules/audio_coding/neteq/interface/webrtc_neteq.h b/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
index 39f6595..9fc8297 100644
--- a/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
+++ b/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
@@ -62,6 +62,7 @@
kDecoderG722_1C_24,
kDecoderG722_1C_32,
kDecoderG722_1C_48,
+ kDecoderOpus,
kDecoderSPEEX_8,
kDecoderSPEEX_16,
kDecoderCELT_32,
diff --git a/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h b/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
index c6f19bb..d885faa 100644
--- a/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
+++ b/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
@@ -151,7 +151,6 @@
inst.funcUpdBWEst=NULL; \
inst.funcGetErrorCode=NULL;
-
#define SET_PCM16B_SWB48_FUNCTIONS(inst) \
inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
inst.funcDecodeRCU=NULL; \
@@ -317,6 +316,17 @@
inst.funcUpdBWEst=NULL; \
inst.funcGetErrorCode=NULL;
+#define SET_OPUS_FUNCTIONS(inst) \
+ inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcOpus_Decode; \
+ inst.funcDecodeRCU=NULL; \
+ inst.funcDecodePLC=NULL; \
+ inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcOpus_DecoderInit; \
+ inst.funcAddLatePkt=NULL; \
+ inst.funcGetMDinfo=NULL; \
+ inst.funcGetPitch=NULL; \
+ inst.funcUpdBWEst=NULL; \
+ inst.funcGetErrorCode=NULL;
+
#define SET_SPEEX_FUNCTIONS(inst) \
inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcSpeex_Decode; \
inst.funcDecodeRCU=NULL; \
diff --git a/src/modules/audio_coding/neteq/neteq_defines.h b/src/modules/audio_coding/neteq/neteq_defines.h
index 318e6bb..79cb144 100644
--- a/src/modules/audio_coding/neteq/neteq_defines.h
+++ b/src/modules/audio_coding/neteq/neteq_defines.h
@@ -77,6 +77,8 @@
*
* NETEQ_G722_1C_CODEC Enable G722.1 Annex C
*
+ * NETEQ_OPUS_CODEC Enable Opus
+ *
* NETEQ_SPEEX_CODEC Enable Speex (at 8 and 16 kHz sample rate)
*
* NETEQ_CELT_CODEC Enable Celt (at 32 kHz sample rate)
@@ -244,6 +246,7 @@
#define NETEQ_G729_CODEC
#define NETEQ_G726_CODEC
#define NETEQ_GSMFR_CODEC
+ #define NETEQ_OPUS_CODEC
#define NETEQ_AMR_CODEC
#endif
@@ -252,6 +255,7 @@
#define NETEQ_G722_CODEC
#define NETEQ_G722_1_CODEC
#define NETEQ_G729_1_CODEC
+ #define NETEQ_OPUS_CODEC
#define NETEQ_SPEEX_CODEC
#define NETEQ_AMRWB_CODEC
#define NETEQ_WIDEBAND
@@ -262,6 +266,7 @@
#define NETEQ_32KHZ_WIDEBAND
#define NETEQ_G722_1C_CODEC
#define NETEQ_CELT_CODEC
+ #define NETEQ_OPUS_CODEC
#endif
#if (defined(NETEQ_VOICEENGINE_CODECS))
@@ -295,6 +300,8 @@
#define NETEQ_G722_1C_CODEC
#define NETEQ_CELT_CODEC
+ /* Fullband 48 kHz codecs */
+ #define NETEQ_OPUS_CODEC
#endif
#if (defined(NETEQ_ALL_CODECS))
@@ -331,21 +338,26 @@
/* Super wideband 48kHz codecs */
#define NETEQ_48KHZ_WIDEBAND
+ #define NETEQ_OPUS_CODEC
#endif
/* Max output size from decoding one frame */
#if defined(NETEQ_48KHZ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 2880 /* 60 ms super wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 3600 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
+ #define NETEQ_MAX_FRAME_SIZE 5760 /* 120 ms super wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 6480 /* 120+15 ms super wideband (120 ms
+ * decoded + 15 ms for merge overlap) */
#elif defined(NETEQ_32KHZ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 1920 /* 60 ms super wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 2400 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
+ #define NETEQ_MAX_FRAME_SIZE 3840 /* 120 ms super wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 4320 /* 120+15 ms super wideband (120 ms
+ * decoded + 15 ms for merge overlap) */
#elif defined(NETEQ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 960 /* 60 ms wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 1200 /* 60+15 ms wideband (60 ms decoded + 10 ms for merge overlap) */
+ #define NETEQ_MAX_FRAME_SIZE 1920 /* 120 ms wideband */
+ #define NETEQ_MAX_OUTPUT_SIZE 2160 /* 120+15 ms wideband (120 ms decoded +
+ * 15 ms for merge overlap) */
#else
- #define NETEQ_MAX_FRAME_SIZE 480 /* 60 ms narrowband */
- #define NETEQ_MAX_OUTPUT_SIZE 600 /* 60+15 ms narrowband (60 ms decoded + 10 ms for merge overlap) */
+ #define NETEQ_MAX_FRAME_SIZE 960 /* 120 ms narrowband */
+ #define NETEQ_MAX_OUTPUT_SIZE 1080 /* 120+15 ms narrowband (120 ms decoded
+ * + 15 ms for merge overlap) */
#endif
diff --git a/src/modules/audio_coding/neteq/packet_buffer.c b/src/modules/audio_coding/neteq/packet_buffer.c
index 8f09b07..7fbea58 100644
--- a/src/modules/audio_coding/neteq/packet_buffer.c
+++ b/src/modules/audio_coding/neteq/packet_buffer.c
@@ -578,6 +578,11 @@
codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */
codecBuffers = 8;
}
+ else if (codecID[i] == kDecoderOpus)
+ {
+ codecBytes = 15300; /* 240ms @ 510kbps (60ms frames) */
+ codecBuffers = 30; /* Replicating the value for PCMu/a */
+ }
else if ((codecID[i] == kDecoderPCM16B) ||
(codecID[i] == kDecoderPCM16B_2ch))
{
diff --git a/src/modules/audio_coding/neteq/recin.c b/src/modules/audio_coding/neteq/recin.c
index bce7c48..399250d 100644
--- a/src/modules/audio_coding/neteq/recin.c
+++ b/src/modules/audio_coding/neteq/recin.c
@@ -202,6 +202,13 @@
/* Get CNG sample rate */
WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
RTPpacket[i_k].payloadType);
+
+ /* Force sampling frequency to 32000 Hz CNG 48000 Hz. */
+ /* TODO(tlegrand): remove limitation once ACM has full 48 kHz
+ * support. */
+ if (fsCng > 32000) {
+ fsCng = 32000;
+ }
if ((fsCng != MCU_inst->fs) && (fsCng > 8000))
{
/*
@@ -370,10 +377,29 @@
MCU_inst->scalingFactor = kTSscalingTwo;
break;
}
+ case kDecoderOpus:
+ {
+ /* We resample Opus internally to 32 kHz, but timestamps
+ * are counted at 48 kHz. So there are two output samples
+ * per three RTP timestamp ticks. */
+ MCU_inst->scalingFactor = kTSscalingTwoThirds;
+ break;
+ }
+
case kDecoderAVT:
case kDecoderCNG:
{
- /* do not change the timestamp scaling settings */
+ /* TODO(tlegrand): remove scaling once ACM has full 48 kHz
+ * support. */
+ WebRtc_UWord16 sample_freq =
+ WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
+ rtpPayloadType);
+ if (sample_freq == 48000) {
+ MCU_inst->scalingFactor = kTSscalingTwoThirds;
+ }
+
+ /* For sample_freq <= 32 kHz, do not change the timestamp scaling
+ * settings. */
break;
}
default:
diff --git a/src/modules/audio_coding/neteq/recout.c b/src/modules/audio_coding/neteq/recout.c
index eb80f2d..1f47945 100644
--- a/src/modules/audio_coding/neteq/recout.c
+++ b/src/modules/audio_coding/neteq/recout.c
@@ -41,8 +41,8 @@
/* Scratch usage:
Type Name size startpos endpos
- WebRtc_Word16 pw16_NetEqAlgorithm_buffer 600*fs/8000 0 600*fs/8000-1
- struct dspInfo 6 600*fs/8000 605*fs/8000
+ WebRtc_Word16 pw16_NetEqAlgorithm_buffer 1080*fs/8000 0 1080*fs/8000-1
+ struct dspInfo 6 1080*fs/8000 1085*fs/8000
func WebRtcNetEQ_Normal 40+495*fs/8000 0 39+495*fs/8000
func WebRtcNetEQ_Merge 40+496*fs/8000 0 39+496*fs/8000
@@ -50,7 +50,7 @@
func WebRtcNetEQ_Accelerate 210 240*fs/8000 209+240*fs/8000
func WebRtcNetEQ_BGNUpdate 69 480*fs/8000 68+480*fs/8000
- Total: 605*fs/8000
+ Total: 1086*fs/8000
*/
#define SCRATCH_ALGORITHM_BUFFER 0
@@ -58,35 +58,35 @@
#define SCRATCH_NETEQ_MERGE 0
#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SCRATCH_DSP_INFO 3600
+#define SCRATCH_DSP_INFO 6480
#define SCRATCH_NETEQ_ACCELERATE 1440
#define SCRATCH_NETEQ_BGN_UPDATE 2880
#define SCRATCH_NETEQ_EXPAND 756
#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SCRATCH_DSP_INFO 2400
+#define SCRATCH_DSP_INFO 4320
#define SCRATCH_NETEQ_ACCELERATE 960
#define SCRATCH_NETEQ_BGN_UPDATE 1920
#define SCRATCH_NETEQ_EXPAND 504
#elif (defined(NETEQ_WIDEBAND))
-#define SCRATCH_DSP_INFO 1200
+#define SCRATCH_DSP_INFO 2160
#define SCRATCH_NETEQ_ACCELERATE 480
#define SCRATCH_NETEQ_BGN_UPDATE 960
#define SCRATCH_NETEQ_EXPAND 252
#else /* NB */
-#define SCRATCH_DSP_INFO 600
+#define SCRATCH_DSP_INFO 1080
#define SCRATCH_NETEQ_ACCELERATE 240
#define SCRATCH_NETEQ_BGN_UPDATE 480
#define SCRATCH_NETEQ_EXPAND 126
#endif
#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 3636
+#define SIZE_SCRATCH_BUFFER 6516
#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 2424
+#define SIZE_SCRATCH_BUFFER 4344
#elif (defined(NETEQ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 1212
+#define SIZE_SCRATCH_BUFFER 2172
#else /* NB */
-#define SIZE_SCRATCH_BUFFER 606
+#define SIZE_SCRATCH_BUFFER 1086
#endif
#ifdef NETEQ_DELAY_LOGGING
@@ -110,13 +110,15 @@
#ifdef SCRATCH
char pw8_ScratchBuffer[((SIZE_SCRATCH_BUFFER + 1) * 2)];
WebRtc_Word16 *pw16_scratchPtr = (WebRtc_Word16*) pw8_ScratchBuffer;
- WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
+ /* pad with 240*fs_mult to match the overflow guard below */
+ WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
WebRtc_Word16 *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr
+ SCRATCH_ALGORITHM_BUFFER;
DSP2MCU_info_t *dspInfo = (DSP2MCU_info_t*) (pw16_scratchPtr + SCRATCH_DSP_INFO);
#else
- WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
- WebRtc_Word16 pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE];
+ /* pad with 240*fs_mult to match the overflow guard below */
+ WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE+240*6];
+ WebRtc_Word16 pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE+240*6];
DSP2MCU_info_t dspInfoStruct;
DSP2MCU_info_t *dspInfo = &dspInfoStruct;
#endif
diff --git a/src/modules/audio_coding/neteq/signal_mcu.c b/src/modules/audio_coding/neteq/signal_mcu.c
index b28f39c..2cccf1a 100644
--- a/src/modules/audio_coding/neteq/signal_mcu.c
+++ b/src/modules/audio_coding/neteq/signal_mcu.c
@@ -319,7 +319,13 @@
WebRtc_UWord16 tempFs;
tempFs = WebRtcNetEQ_DbGetSampleRate(&inst->codec_DB_inst, payloadType);
- if (tempFs > 0)
+ /* TODO(tlegrand): Remove this limitation once ACM has full
+ * 48 kHz support. */
+ if (tempFs > 32000)
+ {
+ inst->fs = 32000;
+ }
+ else if (tempFs > 0)
{
inst->fs = tempFs;
}
diff --git a/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
index 4ece1bf..9ffac2d 100644
--- a/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
@@ -30,6 +30,7 @@
kNbInHz = 8000,
kWbInHz = 16000,
kSwbInHz = 32000,
+ kFbInHz = 48000,
kLowestPossible = -1,
kDefaultFrequency = kWbInHz
};
diff --git a/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 851642c..1fdd9dc 100644
--- a/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -282,6 +282,12 @@
SetOutputFrequency(kSwbInHz);
}
break;
+ case 48000:
+ if(OutputFrequency() != kFbInHz)
+ {
+ SetOutputFrequency(kFbInHz);
+ }
+ break;
default:
assert(false);
diff --git a/src/modules/modules.gyp b/src/modules/modules.gyp
index 089f538..a5b31c6 100644
--- a/src/modules/modules.gyp
+++ b/src/modules/modules.gyp
@@ -15,6 +15,7 @@
'audio_coding/codecs/ilbc/ilbc.gypi',
'audio_coding/codecs/isac/main/source/isac.gypi',
'audio_coding/codecs/isac/fix/source/isacfix.gypi',
+ 'audio_coding/codecs/opus/opus.gypi',
'audio_coding/codecs/pcm16b/pcm16b.gypi',
'audio_coding/main/source/audio_coding_module.gypi',
'audio_coding/neteq/neteq.gypi',
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index a57da75..c9fd3df 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -27,6 +27,7 @@
_cngNBPayloadType(-1),
_cngWBPayloadType(-1),
_cngSWBPayloadType(-1),
+ _cngFBPayloadType(-1),
_cngPayloadType(-1),
_G722PayloadType(-1),
_lastReceivedG722(false),
@@ -94,7 +95,7 @@
RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
WebRtc_UWord32& frequency)
{
- // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
+ // We can have four CNG on 8000Hz, 16000Hz, 32000Hz and 48000Hz.
if(_cngNBPayloadType == payloadType)
{
frequency = 8000;
@@ -129,6 +130,15 @@
}
_cngPayloadType = _cngSWBPayloadType;
return true;
+ }else if(_cngFBPayloadType == payloadType)
+ {
+ frequency = 48000;
+ if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngFBPayloadType))
+ {
+ ResetStatistics();
+ }
+ _cngPayloadType = _cngFBPayloadType;
+ return true;
}else
{
// not CNG
@@ -195,6 +205,8 @@
_cngWBPayloadType = payloadType;
} else if(frequency == 32000) {
_cngSWBPayloadType = payloadType;
+ } else if(frequency == 48000) {
+ _cngFBPayloadType = payloadType;
} else {
assert(false);
return NULL;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 0b0ba30..e256dd1 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -81,6 +81,7 @@
WebRtc_Word8 _cngNBPayloadType;
WebRtc_Word8 _cngWBPayloadType;
WebRtc_Word8 _cngSWBPayloadType;
+ WebRtc_Word8 _cngFBPayloadType;
WebRtc_Word8 _cngPayloadType;
// G722 is special since it use the wrong number of RTP samples in timestamp VS. number of samples in the frame
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 0f6f69f..0c422de 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -38,6 +38,7 @@
_cngNBPayloadType(-1),
_cngWBPayloadType(-1),
_cngSWBPayloadType(-1),
+ _cngFBPayloadType(-1),
_lastPayloadType(-1),
_includeAudioLevelIndication(false), // @TODO - reset at Init()?
_audioLevelIndicationID(0),
@@ -101,6 +102,10 @@
} else if (frequency == 32000) {
_cngSWBPayloadType = payloadType;
+
+ } else if (frequency == 48000) {
+ _cngFBPayloadType = payloadType;
+
} else {
return -1;
}
@@ -159,6 +164,15 @@
return false;
}
}
+ if(_cngFBPayloadType != -1)
+ {
+ // we have configured SWB CNG
+ if(_cngFBPayloadType == payloadType)
+ {
+ // only set a marker bit when we change payload type to a non CNG
+ return false;
+ }
+ }
// payloadType differ
if(_lastPayloadType == -1)
{
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.h b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
index 5974441..fe9a952 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -117,6 +117,7 @@
WebRtc_Word8 _cngNBPayloadType;
WebRtc_Word8 _cngWBPayloadType;
WebRtc_Word8 _cngSWBPayloadType;
+ WebRtc_Word8 _cngFBPayloadType;
WebRtc_Word8 _lastPayloadType;
// Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
diff --git a/src/voice_engine/channel.cc b/src/voice_engine/channel.cc
index b4889e2..4370b74 100644
--- a/src/voice_engine/channel.cc
+++ b/src/voice_engine/channel.cc
@@ -6403,12 +6403,12 @@
}
WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency();
- if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
- {
- if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
- {
- playoutFrequency = 8000;
- }
+ if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0) {
+ if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) {
+ playoutFrequency = 8000;
+ } else if (STR_CASE_CMP("opus", currRecCodec.plname) == 0) {
+ playoutFrequency = 48000;
+ }
}
timestamp -= (delayMS * (playoutFrequency/1000));
@@ -6482,16 +6482,20 @@
rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency();
CodecInst currRecCodec;
- if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
- {
- if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
- {
- // Even though the actual sampling rate for G.722 audio is
- // 16,000 Hz, the RTP clock rate for the G722 payload format is
- // 8,000 Hz because that value was erroneously assigned in
- // RFC 1890 and must remain unchanged for backward compatibility.
- rtpReceiveFrequency = 8000;
- }
+ if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0) {
+ if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) {
+ // Even though the actual sampling rate for G.722 audio is
+ // 16,000 Hz, the RTP clock rate for the G722 payload format is
+ // 8,000 Hz because that value was erroneously assigned in
+ // RFC 1890 and must remain unchanged for backward compatibility.
+ rtpReceiveFrequency = 8000;
+ } else if (STR_CASE_CMP("opus", currRecCodec.plname) == 0) {
+ // We are resampling Opus internally to 32,000 Hz until all our
+ // DSP routines can operate at 48,000 Hz, but the RTP clock
+ // rate for the Opus payload format is standardized to 48,000 Hz,
+ // because that is the maximum supported decoding sampling rate.
+ rtpReceiveFrequency = 48000;
+ }
}
const WebRtc_UWord32 timeStampDiff = timestamp - _playoutTimeStampRTP;
@@ -6499,24 +6503,25 @@
if (timeStampDiff > 0)
{
- switch (rtpReceiveFrequency)
- {
- case 8000:
- timeStampDiffMs = timeStampDiff >> 3;
- break;
- case 16000:
- timeStampDiffMs = timeStampDiff >> 4;
- break;
- case 32000:
- timeStampDiffMs = timeStampDiff >> 5;
- break;
- default:
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::UpdatePacketDelay() invalid sample "
- "rate");
- timeStampDiffMs = 0;
- return -1;
+ switch (rtpReceiveFrequency) {
+ case 8000:
+ timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 3);
+ break;
+ case 16000:
+ timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 4);
+ break;
+ case 32000:
+ timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 5);
+ break;
+ case 48000:
+ timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff / 48);
+ break;
+ default:
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, _channelId),
+ "Channel::UpdatePacketDelay() invalid sample rate");
+ timeStampDiffMs = 0;
+ return -1;
}
if (timeStampDiffMs > 5000)
{
@@ -6539,20 +6544,23 @@
if (sequenceNumber - _previousSequenceNumber == 1)
{
WebRtc_UWord16 packetDelayMs = 0;
- switch (rtpReceiveFrequency)
- {
- case 8000:
- packetDelayMs = (WebRtc_UWord16)(
+ switch (rtpReceiveFrequency) {
+ case 8000:
+ packetDelayMs = static_cast<WebRtc_UWord16>(
(timestamp - _previousTimestamp) >> 3);
break;
- case 16000:
- packetDelayMs = (WebRtc_UWord16)(
+ case 16000:
+ packetDelayMs = static_cast<WebRtc_UWord16>(
(timestamp - _previousTimestamp) >> 4);
break;
- case 32000:
- packetDelayMs = (WebRtc_UWord16)(
+ case 32000:
+ packetDelayMs = static_cast<WebRtc_UWord16>(
(timestamp - _previousTimestamp) >> 5);
break;
+ case 48000:
+ packetDelayMs = static_cast<WebRtc_UWord16>(
+ (timestamp - _previousTimestamp) / 48);
+ break;
}
if (packetDelayMs >= 10 && packetDelayMs <= 60)
diff --git a/src/voice_engine/transmit_mixer.cc b/src/voice_engine/transmit_mixer.cc
index 6153c5b..d987c4e 100644
--- a/src/voice_engine/transmit_mixer.cc
+++ b/src/voice_engine/transmit_mixer.cc
@@ -322,8 +322,14 @@
if (codec.channels == 2)
stereo_codec_ = true;
- if (codec.plfreq > _mixingFrequency)
+
+ // TODO(tlegrand): Remove once we have full 48 kHz support in
+ // Audio Coding Module.
+ if (codec.plfreq > 32000) {
+ _mixingFrequency = 32000;
+ } else if (codec.plfreq > _mixingFrequency) {
_mixingFrequency = codec.plfreq;
+ }
}
channel = sc.GetNextChannel(iterator);
}
diff --git a/third_party/opus/opus.gyp b/third_party/opus/opus.gyp
new file mode 100644
index 0000000..88c826a
--- /dev/null
+++ b/third_party/opus/opus.gyp
@@ -0,0 +1,184 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'opus',
+ 'type': 'static_library',
+ 'defines': [
+ 'OPUS_BUILD',
+ ],
+ 'conditions': [
+ ['OS=="linux"', {
+ 'cflags': [
+ '-std=c99',
+ ],
+ }],
+ ['OS != "win"', {
+ 'defines': [
+ 'HAVE_LRINTF',
+ 'VAR_ARRAYS',
+ ],
+ }],
+ ['OS == "win"', {
+ 'defines': [
+ 'USE_ALLOCA',
+ ['inline','__inline'],
+ ],
+ 'msvs_disabled_warnings':[
+ 4305, # Disable truncation warning in /source/celt/pitch.c[line 80]
+ ],
+ }],
+ ],
+ 'include_dirs': [
+ 'source/celt',
+ 'source/include',
+ 'source/silk',
+ 'source/silk/float',
+ 'source/src',
+ ],
+ 'sources': [
+ # opus wrapper/glue
+ 'source/src/opus.c',
+ 'source/src/opus_decoder.c',
+ 'source/src/opus_encoder.c',
+ 'source/src/repacketizer.c',
+
+ # celt sub-codec
+ 'source/celt/bands.c',
+ 'source/celt/celt.c',
+ 'source/celt/celt_lpc.c',
+ 'source/celt/cwrs.c',
+ 'source/celt/entcode.c',
+ 'source/celt/entdec.c',
+ 'source/celt/entenc.c',
+ 'source/celt/kiss_fft.c',
+ 'source/celt/laplace.c',
+ 'source/celt/mathops.c',
+ 'source/celt/mdct.c',
+ 'source/celt/modes.c',
+ 'source/celt/pitch.c',
+ 'source/celt/quant_bands.c',
+ 'source/celt/rate.c',
+ 'source/celt/vq.c',
+
+ # silk sub-codec
+ 'source/silk/A2NLSF.c',
+ 'source/silk/ana_filt_bank_1.c',
+ 'source/silk/biquad_alt.c',
+ 'source/silk/bwexpander.c',
+ 'source/silk/bwexpander_32.c',
+ 'source/silk/check_control_input.c',
+ 'source/silk/CNG.c',
+ 'source/silk/code_signs.c',
+ 'source/silk/control_audio_bandwidth.c',
+ 'source/silk/control_codec.c',
+ 'source/silk/control_SNR.c',
+ 'source/silk/debug.c',
+ 'source/silk/decode_core.c',
+ 'source/silk/decode_frame.c',
+ 'source/silk/decode_indices.c',
+ 'source/silk/decode_parameters.c',
+ 'source/silk/decode_pitch.c',
+ 'source/silk/decode_pulses.c',
+ 'source/silk/decoder_set_fs.c',
+ 'source/silk/dec_API.c',
+ 'source/silk/enc_API.c',
+ 'source/silk/encode_indices.c',
+ 'source/silk/encode_pulses.c',
+ 'source/silk/gain_quant.c',
+ 'source/silk/HP_variable_cutoff.c',
+ 'source/silk/init_decoder.c',
+ 'source/silk/init_encoder.c',
+ 'source/silk/inner_prod_aligned.c',
+ 'source/silk/interpolate.c',
+ 'source/silk/lin2log.c',
+ 'source/silk/log2lin.c',
+ 'source/silk/LPC_analysis_filter.c',
+ 'source/silk/LPC_inv_pred_gain.c',
+ 'source/silk/LP_variable_cutoff.c',
+ 'source/silk/NLSF2A.c',
+ 'source/silk/NLSF_decode.c',
+ 'source/silk/NLSF_encode.c',
+ 'source/silk/NLSF_del_dec_quant.c',
+ 'source/silk/NLSF_stabilize.c',
+ 'source/silk/NLSF_unpack.c',
+ 'source/silk/NLSF_VQ.c',
+ 'source/silk/NLSF_VQ_weights_laroia.c',
+ 'source/silk/NSQ.c',
+ 'source/silk/NSQ_del_dec.c',
+ 'source/silk/pitch_est_tables.c',
+ 'source/silk/PLC.c',
+ 'source/silk/process_NLSFs.c',
+ 'source/silk/quant_LTP_gains.c',
+ 'source/silk/resampler.c',
+ 'source/silk/resampler_down2.c',
+ 'source/silk/resampler_down2_3.c',
+ 'source/silk/resampler_private_AR2.c',
+ 'source/silk/resampler_private_down_FIR.c',
+ 'source/silk/resampler_private_IIR_FIR.c',
+ 'source/silk/resampler_private_up2_HQ.c',
+ 'source/silk/resampler_rom.c',
+ 'source/silk/shell_coder.c',
+ 'source/silk/sigm_Q15.c',
+ 'source/silk/sort.c',
+ 'source/silk/stereo_decode_pred.c',
+ 'source/silk/stereo_encode_pred.c',
+ 'source/silk/stereo_find_predictor.c',
+ 'source/silk/stereo_LR_to_MS.c',
+ 'source/silk/stereo_MS_to_LR.c',
+ 'source/silk/stereo_quant_pred.c',
+ 'source/silk/sum_sqr_shift.c',
+ 'source/silk/table_LSF_cos.c',
+ 'source/silk/tables_gain.c',
+ 'source/silk/tables_LTP.c',
+ 'source/silk/tables_NLSF_CB_NB_MB.c',
+ 'source/silk/tables_NLSF_CB_WB.c',
+ 'source/silk/tables_other.c',
+ 'source/silk/tables_pitch_lag.c',
+ 'source/silk/tables_pulses_per_block.c',
+ 'source/silk/VAD.c',
+ 'source/silk/VQ_WMat_EC.c',
+
+ # silk floating point engine
+ 'source/silk/float/apply_sine_window_FLP.c',
+ 'source/silk/float/autocorrelation_FLP.c',
+ 'source/silk/float/burg_modified_FLP.c',
+ 'source/silk/float/bwexpander_FLP.c',
+ 'source/silk/float/corrMatrix_FLP.c',
+ 'source/silk/float/encode_frame_FLP.c',
+ 'source/silk/float/energy_FLP.c',
+ 'source/silk/float/find_LPC_FLP.c',
+ 'source/silk/float/find_LTP_FLP.c',
+ 'source/silk/float/find_pitch_lags_FLP.c',
+ 'source/silk/float/find_pred_coefs_FLP.c',
+ 'source/silk/float/inner_product_FLP.c',
+ 'source/silk/float/k2a_FLP.c',
+ 'source/silk/float/levinsondurbin_FLP.c',
+ 'source/silk/float/LPC_analysis_filter_FLP.c',
+ 'source/silk/float/LPC_inv_pred_gain_FLP.c',
+ 'source/silk/float/LTP_analysis_filter_FLP.c',
+ 'source/silk/float/LTP_scale_ctrl_FLP.c',
+ 'source/silk/float/noise_shape_analysis_FLP.c',
+ 'source/silk/float/pitch_analysis_core_FLP.c',
+ 'source/silk/float/prefilter_FLP.c',
+ 'source/silk/float/process_gains_FLP.c',
+ 'source/silk/float/regularize_correlations_FLP.c',
+ 'source/silk/float/residual_energy_FLP.c',
+ 'source/silk/float/scale_copy_vector_FLP.c',
+ 'source/silk/float/scale_vector_FLP.c',
+ 'source/silk/float/schur_FLP.c',
+ 'source/silk/float/solve_LS_FLP.c',
+ 'source/silk/float/sort_FLP.c',
+ 'source/silk/float/warped_autocorrelation_FLP.c',
+ 'source/silk/float/wrappers_FLP.c',
+ ]
+ }
+ ]
+}
diff --git a/tools/create_supplement_gypi.py b/tools/create_supplement_gypi.py
index 7f996e7..f7b2696 100644
--- a/tools/create_supplement_gypi.py
+++ b/tools/create_supplement_gypi.py
@@ -15,6 +15,7 @@
{
'variables': {
'build_with_chromium': 0,
+ 'build_with_mozilla': 0,
}
}
"""
