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<title>[212126] trunk/Source/ThirdParty/libwebrtc</title>
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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/212126">212126</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-02-10 11:03:56 -0800 (Fri, 10 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] Activate libwebrtc G711/G722 audio codecs
https://bugs.webkit.org/show_bug.cgi?id=168123
Patch by Youenn Fablet <youennf@gmail.com> on 2017-02-10
Reviewed by Alex Christensen.
Adding G711/G722 missing codec files.
Activating use of these in the build system.
* Configurations/libwebrtc.xcconfig:
* Source/webrtc/modules/audio_coding/codecs/g711/g711.c: Added.
(ulaw_to_alaw):
* Source/webrtc/modules/audio_coding/codecs/g711/g711.h: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h: Added.
* Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc: Added.
* libwebrtc.xcodeproj/project.pbxproj:</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceThirdPartylibwebrtcChangeLog">trunk/Source/ThirdParty/libwebrtc/ChangeLog</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcConfigurationslibwebrtcxcconfig">trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtclibwebrtcxcodeprojprojectpbxproj">trunk/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj</a></li>
</ul>
<h3>Added Paths</h3>
<ul>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg711g711c">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg711g711h">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h</a></li>
<li>trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/</li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_decoder_g722cc">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_decoder_g722h">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_encoder_g722cc">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_encoder_g722h">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_decodec">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_enc_dech">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_encodec">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_interfacec">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_interfaceh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h</a></li>
<li>trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/</li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722testtestG722cc">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceThirdPartylibwebrtcChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (212125 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-02-10 18:56:07 UTC (rev 212125)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -1,3 +1,29 @@
</span><ins>+2017-02-10 Youenn Fablet <youennf@gmail.com>
+
+ [WebRTC] Activate libwebrtc G711/G722 audio codecs
+ https://bugs.webkit.org/show_bug.cgi?id=168123
+
+ Reviewed by Alex Christensen.
+
+ Adding G711/G722 missing codec files.
+ Activating use of these in the build system.
+
+ * Configurations/libwebrtc.xcconfig:
+ * Source/webrtc/modules/audio_coding/codecs/g711/g711.c: Added.
+ (ulaw_to_alaw):
+ * Source/webrtc/modules/audio_coding/codecs/g711/g711.h: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h: Added.
+ * Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc: Added.
+ * libwebrtc.xcodeproj/project.pbxproj:
+
</ins><span class="cx"> 2017-02-10 Alex Christensen <achristensen@webkit.org>
</span><span class="cx">
</span><span class="cx"> Fix ASAN build.
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcConfigurationslibwebrtcxcconfig"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig (212125 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig 2017-02-10 18:56:07 UTC (rev 212125)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -13,7 +13,7 @@
</span><span class="cx"> USE_HEADERMAP = NO;
</span><span class="cx"> WARNING_CFLAGS = -Wno-exit-time-destructors -Wno-objc-property-no-attribute -Wno-objc-missing-super-calls -Wno-deprecated-declarations;
</span><span class="cx">
</span><del>-GCC_PREPROCESSOR_DEFINITIONS = WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_CODEC_ILBC WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE HAVE_SCTP;
</del><ins>+GCC_PREPROCESSOR_DEFINITIONS = WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_CODEC_ILBC WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722;
</ins><span class="cx">
</span><span class="cx"> GCC_PREPROCESSOR_DEFINITIONS[sdk=macosx*] = $(inherited);
</span><span class="cx"> GCC_PREPROCESSOR_DEFINITIONS[sdk=iphoneos*] = $(inherited) WEBRTC_IOS;
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg711g711c"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,73 @@
</span><ins>+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.c - A-law and u-law transcoding routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place this code in the
+ * public domain for the benefit of all mankind - even the slimy
+ * ones who might try to proprietize my work and use it to my
+ * detriment.
+ *
+ * $Id: g711.c,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed unused include files
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+#include "g711.h"
+#include "webrtc/typedefs.h"
+
+/* Copied from the CCITT G.711 specification */
+static const uint8_t ulaw_to_alaw_table[256] = {
+ 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36,
+ 37, 58, 59, 56, 57, 62, 63, 60, 61, 50, 51, 48, 49, 54, 55,
+ 52, 53, 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6,
+ 7, 4, 26, 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17, 22,
+ 23, 20, 21, 106, 104, 105, 110, 111, 108, 109, 98, 99, 96, 97, 102,
+ 103, 100, 101, 122, 120, 126, 127, 124, 125, 114, 115, 112, 113, 118, 119,
+ 116, 117, 75, 73, 79, 77, 66, 67, 64, 65, 70, 71, 68, 69, 90,
+ 91, 88, 89, 94, 95, 92, 93, 82, 82, 83, 83, 80, 80, 81, 81,
+ 86, 86, 87, 87, 84, 84, 85, 85, 170, 171, 168, 169, 174, 175, 172,
+ 173, 162, 163, 160, 161, 166, 167, 164, 165, 186, 187, 184, 185, 190, 191,
+ 188, 189, 178, 179, 176, 177, 182, 183, 180, 181, 138, 139, 136, 137, 142,
+ 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 154, 155, 152, 153, 158,
+ 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 234, 232, 233, 238,
+ 239, 236, 237, 226, 227, 224, 225, 230, 231, 228, 229, 250, 248, 254, 255,
+ 252, 253, 242, 243, 240, 241, 246, 247, 244, 245, 203, 201, 207, 205, 194,
+ 195, 192, 193, 198, 199, 196, 197, 218, 219, 216, 217, 222, 223, 220, 221,
+ 210, 210, 211, 211, 208, 208, 209, 209, 214, 214, 215, 215, 212, 212, 213,
+ 213
+};
+
+/* These transcoding tables are copied from the CCITT G.711 specification. To
+ achieve optimal results, do not change them. */
+static const uint8_t alaw_to_ulaw_table[256] = {
+ 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36,
+ 37, 57, 58, 55, 56, 61, 62, 59, 60, 49, 50, 47, 48, 53, 54,
+ 51, 52, 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6,
+ 7, 4, 5, 26, 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17,
+ 22, 23, 20, 21, 98, 99, 96, 97, 102, 103, 100, 101, 93, 93, 92,
+ 92, 95, 95, 94, 94, 116, 118, 112, 114, 124, 126, 120, 122, 106, 107,
+ 104, 105, 110, 111, 108, 109, 72, 73, 70, 71, 76, 77, 74, 75, 64,
+ 65, 63, 63, 68, 69, 66, 67, 86, 87, 84, 85, 90, 91, 88, 89,
+ 79, 79, 78, 78, 82, 83, 80, 81, 170, 171, 168, 169, 174, 175, 172,
+ 173, 162, 163, 160, 161, 166, 167, 164, 165, 185, 186, 183, 184, 189, 190,
+ 187, 188, 177, 178, 175, 176, 181, 182, 179, 180, 138, 139, 136, 137, 142,
+ 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 133, 154, 155, 152, 153,
+ 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 226, 227, 224,
+ 225, 230, 231, 228, 229, 221, 221, 220, 220, 223, 223, 222, 222, 244, 246,
+ 240, 242, 252, 254, 248, 250, 234, 235, 232, 233, 238, 239, 236, 237, 200,
+ 201, 198, 199, 204, 205, 202, 203, 192, 193, 191, 191, 196, 197, 194, 195,
+ 214, 215, 212, 213, 218, 219, 216, 217, 207, 207, 206, 206, 210, 211, 208,
+ 209
+};
+
+uint8_t alaw_to_ulaw(uint8_t alaw) { return alaw_to_ulaw_table[alaw]; }
+
+uint8_t ulaw_to_alaw(uint8_t ulaw) { return ulaw_to_alaw_table[ulaw]; }
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg711g711h"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,344 @@
</span><ins>+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.h - In line A-law and u-law conversion routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place this code in the
+ * public domain for the benefit of all mankind - even the slimy
+ * ones who might try to proprietize my work and use it to my
+ * detriment.
+ *
+ * $Id: g711.h,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Two changes to make implementation bitexact with ITU-T reference implementation
+ */
+
+/*! \page g711_page A-law and mu-law handling
+Lookup tables for A-law and u-law look attractive, until you consider the impact
+on the CPU cache. If it causes a substantial area of your processor cache to get
+hit too often, cache sloshing will severely slow things down. The main reason
+these routines are slow in C, is the lack of direct access to the CPU's "find
+the first 1" instruction. A little in-line assembler fixes that, and the
+conversion routines can be faster than lookup tables, in most real world usage.
+A "find the first 1" instruction is available on most modern CPUs, and is a
+much underused feature.
+
+If an assembly language method of bit searching is not available, these routines
+revert to a method that can be a little slow, so the cache thrashing might not
+seem so bad :(
+
+Feel free to submit patches to add fast "find the first 1" support for your own
+favourite processor.
+
+Look up tables are used for transcoding between A-law and u-law, since it is
+difficult to achieve the precise transcoding procedure laid down in the G.711
+specification by other means.
+*/
+
+#if !defined(_G711_H_)
+#define _G711_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "webrtc/typedefs.h"
+
+#if defined(__i386__)
+/*! \brief Find the bit position of the highest set bit in a word
+ \param bits The word to be searched
+ \return The bit number of the highest set bit, or -1 if the word is zero. */
+static __inline__ int top_bit(unsigned int bits) {
+ int res;
+
+ __asm__ __volatile__(" movl $-1,%%edx;\n"
+ " bsrl %%eax,%%edx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+
+/*! \brief Find the bit position of the lowest set bit in a word
+ \param bits The word to be searched
+ \return The bit number of the lowest set bit, or -1 if the word is zero. */
+static __inline__ int bottom_bit(unsigned int bits) {
+ int res;
+
+ __asm__ __volatile__(" movl $-1,%%edx;\n"
+ " bsfl %%eax,%%edx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+#elif defined(__x86_64__)
+static __inline__ int top_bit(unsigned int bits) {
+ int res;
+
+ __asm__ __volatile__(" movq $-1,%%rdx;\n"
+ " bsrq %%rax,%%rdx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+
+static __inline__ int bottom_bit(unsigned int bits) {
+ int res;
+
+ __asm__ __volatile__(" movq $-1,%%rdx;\n"
+ " bsfq %%rax,%%rdx;\n"
+ : "=d" (res)
+ : "a" (bits));
+ return res;
+}
+#else
+static __inline int top_bit(unsigned int bits) {
+ int i;
+
+ if (bits == 0) {
+ return -1;
+ }
+ i = 0;
+ if (bits & 0xFFFF0000) {
+ bits &= 0xFFFF0000;
+ i += 16;
+ }
+ if (bits & 0xFF00FF00) {
+ bits &= 0xFF00FF00;
+ i += 8;
+ }
+ if (bits & 0xF0F0F0F0) {
+ bits &= 0xF0F0F0F0;
+ i += 4;
+ }
+ if (bits & 0xCCCCCCCC) {
+ bits &= 0xCCCCCCCC;
+ i += 2;
+ }
+ if (bits & 0xAAAAAAAA) {
+ bits &= 0xAAAAAAAA;
+ i += 1;
+ }
+ return i;
+}
+
+static __inline int bottom_bit(unsigned int bits) {
+ int i;
+
+ if (bits == 0) {
+ return -1;
+ }
+ i = 32;
+ if (bits & 0x0000FFFF) {
+ bits &= 0x0000FFFF;
+ i -= 16;
+ }
+ if (bits & 0x00FF00FF) {
+ bits &= 0x00FF00FF;
+ i -= 8;
+ }
+ if (bits & 0x0F0F0F0F) {
+ bits &= 0x0F0F0F0F;
+ i -= 4;
+ }
+ if (bits & 0x33333333) {
+ bits &= 0x33333333;
+ i -= 2;
+ }
+ if (bits & 0x55555555) {
+ bits &= 0x55555555;
+ i -= 1;
+ }
+ return i;
+}
+#endif
+
+/* N.B. It is tempting to use look-up tables for A-law and u-law conversion.
+ * However, you should consider the cache footprint.
+ *
+ * A 64K byte table for linear to x-law and a 512 byte table for x-law to
+ * linear sound like peanuts these days, and shouldn't an array lookup be
+ * real fast? No! When the cache sloshes as badly as this one will, a tight
+ * calculation may be better. The messiest part is normally finding the
+ * segment, but a little inline assembly can fix that on an i386, x86_64 and
+ * many other modern processors.
+ */
+
+/*
+ * Mu-law is basically as follows:
+ *
+ * Biased Linear Input Code Compressed Code
+ * ------------------------ ---------------
+ * 00000001wxyza 000wxyz
+ * 0000001wxyzab 001wxyz
+ * 000001wxyzabc 010wxyz
+ * 00001wxyzabcd 011wxyz
+ * 0001wxyzabcde 100wxyz
+ * 001wxyzabcdef 101wxyz
+ * 01wxyzabcdefg 110wxyz
+ * 1wxyzabcdefgh 111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz. * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+//#define ULAW_ZEROTRAP /* turn on the trap as per the MIL-STD */
+#define ULAW_BIAS 0x84 /* Bias for linear code. */
+
+/*! \brief Encode a linear sample to u-law
+ \param linear The sample to encode.
+ \return The u-law value.
+*/
+static __inline uint8_t linear_to_ulaw(int linear) {
+ uint8_t u_val;
+ int mask;
+ int seg;
+
+ /* Get the sign and the magnitude of the value. */
+ if (linear < 0) {
+ /* WebRtc, tlegrand: -1 added to get bitexact to reference implementation */
+ linear = ULAW_BIAS - linear - 1;
+ mask = 0x7F;
+ } else {
+ linear = ULAW_BIAS + linear;
+ mask = 0xFF;
+ }
+
+ seg = top_bit(linear | 0xFF) - 7;
+
+ /*
+ * Combine the sign, segment, quantization bits,
+ * and complement the code word.
+ */
+ if (seg >= 8)
+ u_val = (uint8_t)(0x7F ^ mask);
+ else
+ u_val = (uint8_t)(((seg << 4) | ((linear >> (seg + 3)) & 0xF)) ^ mask);
+#ifdef ULAW_ZEROTRAP
+ /* Optional ITU trap */
+ if (u_val == 0)
+ u_val = 0x02;
+#endif
+ return u_val;
+}
+
+/*! \brief Decode an u-law sample to a linear value.
+ \param ulaw The u-law sample to decode.
+ \return The linear value.
+*/
+static __inline int16_t ulaw_to_linear(uint8_t ulaw) {
+ int t;
+
+ /* Complement to obtain normal u-law value. */
+ ulaw = ~ulaw;
+ /*
+ * Extract and bias the quantization bits. Then
+ * shift up by the segment number and subtract out the bias.
+ */
+ t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int) ulaw & 0x70) >> 4);
+ return (int16_t)((ulaw & 0x80) ? (ULAW_BIAS - t) : (t - ULAW_BIAS));
+}
+
+/*
+ * A-law is basically as follows:
+ *
+ * Linear Input Code Compressed Code
+ * ----------------- ---------------
+ * 0000000wxyza 000wxyz
+ * 0000001wxyza 001wxyz
+ * 000001wxyzab 010wxyz
+ * 00001wxyzabc 011wxyz
+ * 0001wxyzabcd 100wxyz
+ * 001wxyzabcde 101wxyz
+ * 01wxyzabcdef 110wxyz
+ * 1wxyzabcdefg 111wxyz
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+#define ALAW_AMI_MASK 0x55
+
+/*! \brief Encode a linear sample to A-law
+ \param linear The sample to encode.
+ \return The A-law value.
+*/
+static __inline uint8_t linear_to_alaw(int linear) {
+ int mask;
+ int seg;
+
+ if (linear >= 0) {
+ /* Sign (bit 7) bit = 1 */
+ mask = ALAW_AMI_MASK | 0x80;
+ } else {
+ /* Sign (bit 7) bit = 0 */
+ mask = ALAW_AMI_MASK;
+ /* WebRtc, tlegrand: Changed from -8 to -1 to get bitexact to reference
+ * implementation */
+ linear = -linear - 1;
+ }
+
+ /* Convert the scaled magnitude to segment number. */
+ seg = top_bit(linear | 0xFF) - 7;
+ if (seg >= 8) {
+ if (linear >= 0) {
+ /* Out of range. Return maximum value. */
+ return (uint8_t)(0x7F ^ mask);
+ }
+ /* We must be just a tiny step below zero */
+ return (uint8_t)(0x00 ^ mask);
+ }
+ /* Combine the sign, segment, and quantization bits. */
+ return (uint8_t)(((seg << 4) | ((linear >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^
+ mask);
+}
+
+/*! \brief Decode an A-law sample to a linear value.
+ \param alaw The A-law sample to decode.
+ \return The linear value.
+*/
+static __inline int16_t alaw_to_linear(uint8_t alaw) {
+ int i;
+ int seg;
+
+ alaw ^= ALAW_AMI_MASK;
+ i = ((alaw & 0x0F) << 4);
+ seg = (((int) alaw & 0x70) >> 4);
+ if (seg)
+ i = (i + 0x108) << (seg - 1);
+ else
+ i += 8;
+ return (int16_t)((alaw & 0x80) ? i : -i);
+}
+
+/*! \brief Transcode from A-law to u-law, using the procedure defined in G.711.
+ \param alaw The A-law sample to transcode.
+ \return The best matching u-law value.
+*/
+uint8_t alaw_to_ulaw(uint8_t alaw);
+
+/*! \brief Transcode from u-law to A-law, using the procedure defined in G.711.
+ \param alaw The u-law sample to transcode.
+ \return The best matching A-law value.
+*/
+uint8_t ulaw_to_alaw(uint8_t ulaw);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_decoder_g722cc"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,161 @@
</span><ins>+/*
+ * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+namespace webrtc {
+
+AudioDecoderG722::AudioDecoderG722() {
+ WebRtcG722_CreateDecoder(&dec_state_);
+ WebRtcG722_DecoderInit(dec_state_);
+}
+
+AudioDecoderG722::~AudioDecoderG722() {
+ WebRtcG722_FreeDecoder(dec_state_);
+}
+
+bool AudioDecoderG722::HasDecodePlc() const {
+ return false;
+}
+
+int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) {
+ RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+ int16_t temp_type = 1; // Default is speech.
+ size_t ret =
+ WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+ *speech_type = ConvertSpeechType(temp_type);
+ return static_cast<int>(ret);
+}
+
+void AudioDecoderG722::Reset() {
+ WebRtcG722_DecoderInit(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722::ParsePayload(
+ rtc::Buffer&& payload,
+ uint32_t timestamp) {
+ return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+ timestamp, 8, 16);
+}
+
+int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
+ size_t encoded_len) const {
+ // 1/2 encoded byte per sample per channel.
+ return static_cast<int>(2 * encoded_len / Channels());
+}
+
+int AudioDecoderG722::SampleRateHz() const {
+ return 16000;
+}
+
+size_t AudioDecoderG722::Channels() const {
+ return 1;
+}
+
+AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
+ WebRtcG722_CreateDecoder(&dec_state_left_);
+ WebRtcG722_CreateDecoder(&dec_state_right_);
+ WebRtcG722_DecoderInit(dec_state_left_);
+ WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
+ WebRtcG722_FreeDecoder(dec_state_left_);
+ WebRtcG722_FreeDecoder(dec_state_right_);
+}
+
+int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) {
+ RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+ int16_t temp_type = 1; // Default is speech.
+ // De-interleave the bit-stream into two separate payloads.
+ uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
+ SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
+ // Decode left and right.
+ size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
+ encoded_len / 2, decoded, &temp_type);
+ size_t ret = WebRtcG722_Decode(
+ dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
+ encoded_len / 2, &decoded[decoded_len], &temp_type);
+ if (ret == decoded_len) {
+ ret += decoded_len; // Return total number of samples.
+ // Interleave output.
+ for (size_t k = ret / 2; k < ret; k++) {
+ int16_t temp = decoded[k];
+ memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
+ (ret - k - 1) * sizeof(int16_t));
+ decoded[2 * k - ret + 1] = temp;
+ }
+ }
+ *speech_type = ConvertSpeechType(temp_type);
+ delete[] encoded_deinterleaved;
+ return static_cast<int>(ret);
+}
+
+int AudioDecoderG722Stereo::SampleRateHz() const {
+ return 16000;
+}
+
+size_t AudioDecoderG722Stereo::Channels() const {
+ return 2;
+}
+
+void AudioDecoderG722Stereo::Reset() {
+ WebRtcG722_DecoderInit(dec_state_left_);
+ WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Stereo::ParsePayload(
+ rtc::Buffer&& payload,
+ uint32_t timestamp) {
+ return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+ timestamp, 2 * 8, 16);
+}
+
+// Split the stereo packet and place left and right channel after each other
+// in the output array.
+void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
+ size_t encoded_len,
+ uint8_t* encoded_deinterleaved) {
+ // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
+ // where "lx" is 4 bits representing left sample number x, and "rx" right
+ // sample. Two samples fit in one byte, represented with |...|.
+ for (size_t i = 0; i + 1 < encoded_len; i += 2) {
+ uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
+ encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
+ encoded_deinterleaved[i + 1] = right_byte;
+ }
+
+ // Move one byte representing right channel each loop, and place it at the
+ // end of the bytestream vector. After looping the data is reordered to:
+ // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
+ // where N is the total number of samples.
+ for (size_t i = 0; i < encoded_len / 2; i++) {
+ uint8_t right_byte = encoded_deinterleaved[i + 1];
+ memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
+ encoded_len - i - 2);
+ encoded_deinterleaved[encoded_len - 1] = right_byte;
+ }
+}
+
+} // namespace webrtc
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_decoder_g722h"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,79 @@
</span><ins>+/*
+ * Copyright (c) 2015 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_G722_AUDIO_DECODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722 final : public AudioDecoder {
+ public:
+ AudioDecoderG722();
+ ~AudioDecoderG722() override;
+ bool HasDecodePlc() const override;
+ void Reset() override;
+ std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp) override;
+ int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+ int SampleRateHz() const override;
+ size_t Channels() const override;
+
+ protected:
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ G722DecInst* dec_state_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
+};
+
+class AudioDecoderG722Stereo final : public AudioDecoder {
+ public:
+ AudioDecoderG722Stereo();
+ ~AudioDecoderG722Stereo() override;
+ void Reset() override;
+ std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+ uint32_t timestamp) override;
+ int SampleRateHz() const override;
+ size_t Channels() const override;
+
+ protected:
+ int DecodeInternal(const uint8_t* encoded,
+ size_t encoded_len,
+ int sample_rate_hz,
+ int16_t* decoded,
+ SpeechType* speech_type) override;
+
+ private:
+ // Splits the stereo-interleaved payload in |encoded| into separate payloads
+ // for left and right channels. The separated payloads are written to
+ // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
+ // The left channel starts at offset 0, while the right channel starts at
+ // offset encoded_len / 2 into |encoded_deinterleaved|.
+ void SplitStereoPacket(const uint8_t* encoded,
+ size_t encoded_len,
+ uint8_t* encoded_deinterleaved);
+
+ G722DecInst* dec_state_left_;
+ G722DecInst* dec_state_right_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_encoder_g722cc"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,164 @@
</span><ins>+/*
+ * Copyright (c) 2014 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 "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+
+#include <limits>
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 16000;
+
+AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
+ AudioEncoderG722::Config config;
+ config.num_channels = codec_inst.channels;
+ config.frame_size_ms = codec_inst.pacsize / 16;
+ config.payload_type = codec_inst.pltype;
+ return config;
+}
+
+} // namespace
+
+bool AudioEncoderG722::Config::IsOk() const {
+ return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
+ (num_channels >= 1);
+}
+
+AudioEncoderG722::AudioEncoderG722(const Config& config)
+ : num_channels_(config.num_channels),
+ payload_type_(config.payload_type),
+ num_10ms_frames_per_packet_(
+ static_cast<size_t>(config.frame_size_ms / 10)),
+ num_10ms_frames_buffered_(0),
+ first_timestamp_in_buffer_(0),
+ encoders_(new EncoderState[num_channels_]),
+ interleave_buffer_(2 * num_channels_) {
+ RTC_CHECK(config.IsOk());
+ const size_t samples_per_channel =
+ kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+ for (size_t i = 0; i < num_channels_; ++i) {
+ encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
+ encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
+ }
+ Reset();
+}
+
+AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
+ : AudioEncoderG722(CreateConfig(codec_inst)) {}
+
+AudioEncoderG722::~AudioEncoderG722() = default;
+
+int AudioEncoderG722::SampleRateHz() const {
+ return kSampleRateHz;
+}
+
+size_t AudioEncoderG722::NumChannels() const {
+ return num_channels_;
+}
+
+int AudioEncoderG722::RtpTimestampRateHz() const {
+ // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
+ // codec.
+ return kSampleRateHz / 2;
+}
+
+size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
+ return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
+ return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderG722::GetTargetBitrate() const {
+ // 4 bits/sample, 16000 samples/s/channel.
+ return static_cast<int>(64000 * NumChannels());
+}
+
+void AudioEncoderG722::Reset() {
+ num_10ms_frames_buffered_ = 0;
+ for (size_t i = 0; i < num_channels_; ++i)
+ RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
+ uint32_t rtp_timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ rtc::Buffer* encoded) {
+ if (num_10ms_frames_buffered_ == 0)
+ first_timestamp_in_buffer_ = rtp_timestamp;
+
+ // Deinterleave samples and save them in each channel's buffer.
+ const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+ for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+ for (size_t j = 0; j < num_channels_; ++j)
+ encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
+
+ // If we don't yet have enough samples for a packet, we're done for now.
+ if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+ return EncodedInfo();
+ }
+
+ // Encode each channel separately.
+ RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+ num_10ms_frames_buffered_ = 0;
+ const size_t samples_per_channel = SamplesPerChannel();
+ for (size_t i = 0; i < num_channels_; ++i) {
+ const size_t bytes_encoded = WebRtcG722_Encode(
+ encoders_[i].encoder, encoders_[i].speech_buffer.get(),
+ samples_per_channel, encoders_[i].encoded_buffer.data());
+ RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
+ }
+
+ const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
+ EncodedInfo info;
+ info.encoded_bytes = encoded->AppendData(
+ bytes_to_encode, [&] (rtc::ArrayView<uint8_t> encoded) {
+ // Interleave the encoded bytes of the different channels. Each separate
+ // channel and the interleaved stream encodes two samples per byte, most
+ // significant half first.
+ for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+ for (size_t j = 0; j < num_channels_; ++j) {
+ uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+ interleave_buffer_.data()[j] = two_samples >> 4;
+ interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+ }
+ for (size_t j = 0; j < num_channels_; ++j)
+ encoded[i * num_channels_ + j] =
+ interleave_buffer_.data()[2 * j] << 4 |
+ interleave_buffer_.data()[2 * j + 1];
+ }
+
+ return bytes_to_encode;
+ });
+ info.encoded_timestamp = first_timestamp_in_buffer_;
+ info.payload_type = payload_type_;
+ info.encoder_type = CodecType::kG722;
+ return info;
+}
+
+AudioEncoderG722::EncoderState::EncoderState() {
+ RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722::EncoderState::~EncoderState() {
+ RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722::SamplesPerChannel() const {
+ return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+}
+
+} // namespace webrtc
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722audio_encoder_g722h"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,75 @@
</span><ins>+/*
+ * Copyright (c) 2014 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_G722_AUDIO_ENCODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+namespace webrtc {
+
+struct CodecInst;
+
+class AudioEncoderG722 final : public AudioEncoder {
+ public:
+ struct Config {
+ bool IsOk() const;
+
+ int payload_type = 9;
+ int frame_size_ms = 20;
+ size_t num_channels = 1;
+ };
+
+ explicit AudioEncoderG722(const Config& config);
+ explicit AudioEncoderG722(const CodecInst& codec_inst);
+ ~AudioEncoderG722() override;
+
+ int SampleRateHz() const override;
+ size_t NumChannels() const override;
+ int RtpTimestampRateHz() const override;
+ size_t Num10MsFramesInNextPacket() const override;
+ size_t Max10MsFramesInAPacket() const override;
+ int GetTargetBitrate() const override;
+ void Reset() override;
+
+ protected:
+ EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ rtc::Buffer* encoded) override;
+
+ private:
+ // The encoder state for one channel.
+ struct EncoderState {
+ G722EncInst* encoder;
+ std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
+ rtc::Buffer encoded_buffer; // Already encoded.
+ EncoderState();
+ ~EncoderState();
+ };
+
+ size_t SamplesPerChannel() const;
+
+ const size_t num_channels_;
+ const int payload_type_;
+ const size_t num_10ms_frames_per_packet_;
+ size_t num_10ms_frames_buffered_;
+ uint32_t first_timestamp_in_buffer_;
+ const std::unique_ptr<EncoderState[]> encoders_;
+ rtc::Buffer interleave_buffer_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_decodec"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,400 @@
</span><ins>+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_decode.c - The ITU G.722 codec, decode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based in part on a single channel G.722 codec which is:
+ *
+ * Copyright (c) CMU 1993
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_decode.c,v 1.15 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Added saturation check on output
+ */
+
+/*! \file */
+
+
+#include <memory.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "g722_enc_dec.h"
+#include "webrtc/typedefs.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline int16_t saturate(int32_t amp)
+{
+ int16_t amp16;
+
+ /* Hopefully this is optimised for the common case - not clipping */
+ amp16 = (int16_t) amp;
+ if (amp == amp16)
+ return amp16;
+ if (amp > WEBRTC_INT16_MAX)
+ return WEBRTC_INT16_MAX;
+ return WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(G722DecoderState *s, int band, int d);
+
+static void block4(G722DecoderState *s, int band, int d)
+{
+ int wd1;
+ int wd2;
+ int wd3;
+ int i;
+
+ /* Block 4, RECONS */
+ s->band[band].d[0] = d;
+ s->band[band].r[0] = saturate(s->band[band].s + d);
+
+ /* Block 4, PARREC */
+ s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+ /* Block 4, UPPOL2 */
+ for (i = 0; i < 3; i++)
+ s->band[band].sg[i] = s->band[band].p[i] >> 15;
+ wd1 = saturate(s->band[band].a[1] * 4);
+
+ wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
+ if (wd2 > 32767)
+ wd2 = 32767;
+ wd3 = (s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128;
+ wd3 += (wd2 >> 7);
+ wd3 += (s->band[band].a[2]*32512) >> 15;
+ if (wd3 > 12288)
+ wd3 = 12288;
+ else if (wd3 < -12288)
+ wd3 = -12288;
+ s->band[band].ap[2] = wd3;
+
+ /* Block 4, UPPOL1 */
+ s->band[band].sg[0] = s->band[band].p[0] >> 15;
+ s->band[band].sg[1] = s->band[band].p[1] >> 15;
+ wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
+ wd2 = (s->band[band].a[1]*32640) >> 15;
+
+ s->band[band].ap[1] = saturate(wd1 + wd2);
+ wd3 = saturate(15360 - s->band[band].ap[2]);
+ if (s->band[band].ap[1] > wd3)
+ s->band[band].ap[1] = wd3;
+ else if (s->band[band].ap[1] < -wd3)
+ s->band[band].ap[1] = -wd3;
+
+ /* Block 4, UPZERO */
+ wd1 = (d == 0) ? 0 : 128;
+ s->band[band].sg[0] = d >> 15;
+ for (i = 1; i < 7; i++)
+ {
+ s->band[band].sg[i] = s->band[band].d[i] >> 15;
+ wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
+ wd3 = (s->band[band].b[i]*32640) >> 15;
+ s->band[band].bp[i] = saturate(wd2 + wd3);
+ }
+
+ /* Block 4, DELAYA */
+ for (i = 6; i > 0; i--)
+ {
+ s->band[band].d[i] = s->band[band].d[i - 1];
+ s->band[band].b[i] = s->band[band].bp[i];
+ }
+
+ for (i = 2; i > 0; i--)
+ {
+ s->band[band].r[i] = s->band[band].r[i - 1];
+ s->band[band].p[i] = s->band[band].p[i - 1];
+ s->band[band].a[i] = s->band[band].ap[i];
+ }
+
+ /* Block 4, FILTEP */
+ wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+ wd1 = (s->band[band].a[1]*wd1) >> 15;
+ wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+ wd2 = (s->band[band].a[2]*wd2) >> 15;
+ s->band[band].sp = saturate(wd1 + wd2);
+
+ /* Block 4, FILTEZ */
+ s->band[band].sz = 0;
+ for (i = 6; i > 0; i--)
+ {
+ wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+ s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+ }
+ s->band[band].sz = saturate(s->band[band].sz);
+
+ /* Block 4, PREDIC */
+ s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+ int rate,
+ int options) {
+ s = s ? s : malloc(sizeof(*s));
+ memset(s, 0, sizeof(*s));
+ if (rate == 48000)
+ s->bits_per_sample = 6;
+ else if (rate == 56000)
+ s->bits_per_sample = 7;
+ else
+ s->bits_per_sample = 8;
+ if ((options & G722_SAMPLE_RATE_8000))
+ s->eight_k = TRUE;
+ if ((options & G722_PACKED) && s->bits_per_sample != 8)
+ s->packed = TRUE;
+ else
+ s->packed = FALSE;
+ s->band[0].det = 32;
+ s->band[1].det = 8;
+ return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_decode_release(G722DecoderState *s)
+{
+ free(s);
+ return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+size_t WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
+ const uint8_t g722_data[], size_t len)
+{
+ static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
+ static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
+ 7, 6, 5, 4, 3, 2, 1, 0 };
+ static const int ilb[32] =
+ {
+ 2048, 2093, 2139, 2186, 2233, 2282, 2332,
+ 2383, 2435, 2489, 2543, 2599, 2656, 2714,
+ 2774, 2834, 2896, 2960, 3025, 3091, 3158,
+ 3228, 3298, 3371, 3444, 3520, 3597, 3676,
+ 3756, 3838, 3922, 4008
+ };
+ static const int wh[3] = {0, -214, 798};
+ static const int rh2[4] = {2, 1, 2, 1};
+ static const int qm2[4] = {-7408, -1616, 7408, 1616};
+ static const int qm4[16] =
+ {
+ 0, -20456, -12896, -8968,
+ -6288, -4240, -2584, -1200,
+ 20456, 12896, 8968, 6288,
+ 4240, 2584, 1200, 0
+ };
+ static const int qm5[32] =
+ {
+ -280, -280, -23352, -17560,
+ -14120, -11664, -9752, -8184,
+ -6864, -5712, -4696, -3784,
+ -2960, -2208, -1520, -880,
+ 23352, 17560, 14120, 11664,
+ 9752, 8184, 6864, 5712,
+ 4696, 3784, 2960, 2208,
+ 1520, 880, 280, -280
+ };
+ static const int qm6[64] =
+ {
+ -136, -136, -136, -136,
+ -24808, -21904, -19008, -16704,
+ -14984, -13512, -12280, -11192,
+ -10232, -9360, -8576, -7856,
+ -7192, -6576, -6000, -5456,
+ -4944, -4464, -4008, -3576,
+ -3168, -2776, -2400, -2032,
+ -1688, -1360, -1040, -728,
+ 24808, 21904, 19008, 16704,
+ 14984, 13512, 12280, 11192,
+ 10232, 9360, 8576, 7856,
+ 7192, 6576, 6000, 5456,
+ 4944, 4464, 4008, 3576,
+ 3168, 2776, 2400, 2032,
+ 1688, 1360, 1040, 728,
+ 432, 136, -432, -136
+ };
+ static const int qmf_coeffs[12] =
+ {
+ 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
+ };
+
+ int dlowt;
+ int rlow;
+ int ihigh;
+ int dhigh;
+ int rhigh;
+ int xout1;
+ int xout2;
+ int wd1;
+ int wd2;
+ int wd3;
+ int code;
+ size_t outlen;
+ int i;
+ size_t j;
+
+ outlen = 0;
+ rhigh = 0;
+ for (j = 0; j < len; )
+ {
+ if (s->packed)
+ {
+ /* Unpack the code bits */
+ if (s->in_bits < s->bits_per_sample)
+ {
+ s->in_buffer |= (g722_data[j++] << s->in_bits);
+ s->in_bits += 8;
+ }
+ code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
+ s->in_buffer >>= s->bits_per_sample;
+ s->in_bits -= s->bits_per_sample;
+ }
+ else
+ {
+ code = g722_data[j++];
+ }
+
+ switch (s->bits_per_sample)
+ {
+ default:
+ case 8:
+ wd1 = code & 0x3F;
+ ihigh = (code >> 6) & 0x03;
+ wd2 = qm6[wd1];
+ wd1 >>= 2;
+ break;
+ case 7:
+ wd1 = code & 0x1F;
+ ihigh = (code >> 5) & 0x03;
+ wd2 = qm5[wd1];
+ wd1 >>= 1;
+ break;
+ case 6:
+ wd1 = code & 0x0F;
+ ihigh = (code >> 4) & 0x03;
+ wd2 = qm4[wd1];
+ break;
+ }
+ /* Block 5L, LOW BAND INVQBL */
+ wd2 = (s->band[0].det*wd2) >> 15;
+ /* Block 5L, RECONS */
+ rlow = s->band[0].s + wd2;
+ /* Block 6L, LIMIT */
+ if (rlow > 16383)
+ rlow = 16383;
+ else if (rlow < -16384)
+ rlow = -16384;
+
+ /* Block 2L, INVQAL */
+ wd2 = qm4[wd1];
+ dlowt = (s->band[0].det*wd2) >> 15;
+
+ /* Block 3L, LOGSCL */
+ wd2 = rl42[wd1];
+ wd1 = (s->band[0].nb*127) >> 7;
+ wd1 += wl[wd2];
+ if (wd1 < 0)
+ wd1 = 0;
+ else if (wd1 > 18432)
+ wd1 = 18432;
+ s->band[0].nb = wd1;
+
+ /* Block 3L, SCALEL */
+ wd1 = (s->band[0].nb >> 6) & 31;
+ wd2 = 8 - (s->band[0].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[0].det = wd3 << 2;
+
+ block4(s, 0, dlowt);
+
+ if (!s->eight_k)
+ {
+ /* Block 2H, INVQAH */
+ wd2 = qm2[ihigh];
+ dhigh = (s->band[1].det*wd2) >> 15;
+ /* Block 5H, RECONS */
+ rhigh = dhigh + s->band[1].s;
+ /* Block 6H, LIMIT */
+ if (rhigh > 16383)
+ rhigh = 16383;
+ else if (rhigh < -16384)
+ rhigh = -16384;
+
+ /* Block 2H, INVQAH */
+ wd2 = rh2[ihigh];
+ wd1 = (s->band[1].nb*127) >> 7;
+ wd1 += wh[wd2];
+ if (wd1 < 0)
+ wd1 = 0;
+ else if (wd1 > 22528)
+ wd1 = 22528;
+ s->band[1].nb = wd1;
+
+ /* Block 3H, SCALEH */
+ wd1 = (s->band[1].nb >> 6) & 31;
+ wd2 = 10 - (s->band[1].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[1].det = wd3 << 2;
+
+ block4(s, 1, dhigh);
+ }
+
+ if (s->itu_test_mode)
+ {
+ amp[outlen++] = (int16_t) (rlow << 1);
+ amp[outlen++] = (int16_t) (rhigh << 1);
+ }
+ else
+ {
+ if (s->eight_k)
+ {
+ amp[outlen++] = (int16_t) (rlow << 1);
+ }
+ else
+ {
+ /* Apply the receive QMF */
+ for (i = 0; i < 22; i++)
+ s->x[i] = s->x[i + 2];
+ s->x[22] = rlow + rhigh;
+ s->x[23] = rlow - rhigh;
+
+ xout1 = 0;
+ xout2 = 0;
+ for (i = 0; i < 12; i++)
+ {
+ xout2 += s->x[2*i]*qmf_coeffs[i];
+ xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
+ }
+ /* We shift by 12 to allow for the QMF filters (DC gain = 4096), less 1
+ to allow for the 15 bit input to the G.722 algorithm. */
+ /* WebRtc, tlegrand: added saturation */
+ amp[outlen++] = saturate(xout1 >> 11);
+ amp[outlen++] = saturate(xout2 >> 11);
+ }
+ }
+ }
+ return outlen;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_enc_dech"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,160 @@
</span><ins>+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722.h - The ITU G.722 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based on a single channel G.722 codec which is:
+ *
+ ***** Copyright (c) CMU 1993 *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Added new defines for minimum and maximum values of short int
+ */
+
+
+/*! \file */
+
+#if !defined(_G722_ENC_DEC_H_)
+#define _G722_ENC_DEC_H_
+
+#include "webrtc/typedefs.h"
+
+/*! \page g722_page G.722 encoding and decoding
+\section g722_page_sec_1 What does it do?
+The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
+specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
+
+To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
+support an option for the linear audio to be an 8k samples/second stream. In this mode the
+codec is considerably faster, and still fully compatible with wideband terminals using G.722.
+
+\section g722_page_sec_2 How does it work?
+???.
+*/
+
+#define WEBRTC_INT16_MAX 32767
+#define WEBRTC_INT16_MIN -32768
+
+enum
+{
+ G722_SAMPLE_RATE_8000 = 0x0001,
+ G722_PACKED = 0x0002
+};
+
+typedef struct
+{
+ /*! TRUE if the operating in the special ITU test mode, with the band split filters
+ disabled. */
+ int itu_test_mode;
+ /*! TRUE if the G.722 data is packed */
+ int packed;
+ /*! TRUE if encode from 8k samples/second */
+ int eight_k;
+ /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+ int bits_per_sample;
+
+ /*! Signal history for the QMF */
+ int x[24];
+
+ struct
+ {
+ int s;
+ int sp;
+ int sz;
+ int r[3];
+ int a[3];
+ int ap[3];
+ int p[3];
+ int d[7];
+ int b[7];
+ int bp[7];
+ int sg[7];
+ int nb;
+ int det;
+ } band[2];
+
+ unsigned int in_buffer;
+ int in_bits;
+ unsigned int out_buffer;
+ int out_bits;
+} G722EncoderState;
+
+typedef struct
+{
+ /*! TRUE if the operating in the special ITU test mode, with the band split filters
+ disabled. */
+ int itu_test_mode;
+ /*! TRUE if the G.722 data is packed */
+ int packed;
+ /*! TRUE if decode to 8k samples/second */
+ int eight_k;
+ /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+ int bits_per_sample;
+
+ /*! Signal history for the QMF */
+ int x[24];
+
+ struct
+ {
+ int s;
+ int sp;
+ int sz;
+ int r[3];
+ int a[3];
+ int ap[3];
+ int p[3];
+ int d[7];
+ int b[7];
+ int bp[7];
+ int sg[7];
+ int nb;
+ int det;
+ } band[2];
+
+ unsigned int in_buffer;
+ int in_bits;
+ unsigned int out_buffer;
+ int out_bits;
+} G722DecoderState;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+ int rate,
+ int options);
+int WebRtc_g722_encode_release(G722EncoderState *s);
+size_t WebRtc_g722_encode(G722EncoderState *s,
+ uint8_t g722_data[],
+ const int16_t amp[],
+ size_t len);
+
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+ int rate,
+ int options);
+int WebRtc_g722_decode_release(G722DecoderState *s);
+size_t WebRtc_g722_decode(G722DecoderState *s,
+ int16_t amp[],
+ const uint8_t g722_data[],
+ size_t len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_encodec"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,430 @@
</span><ins>+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_encode.c - The ITU G.722 codec, encode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * Despite my general liking of the GPL, I place my own contributions
+ * to this code in the public domain for the benefit of all mankind -
+ * even the slimy ones who might try to proprietize my work and use it
+ * to my detriment.
+ *
+ * Based on a single channel 64kbps only G.722 codec which is:
+ *
+ ***** Copyright (c) CMU 1993 *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_encode.c,v 1.14 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+/*! \file */
+
+#include <memory.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "g722_enc_dec.h"
+#include "webrtc/typedefs.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline int16_t saturate(int32_t amp)
+{
+ int16_t amp16;
+
+ /* Hopefully this is optimised for the common case - not clipping */
+ amp16 = (int16_t) amp;
+ if (amp == amp16)
+ return amp16;
+ if (amp > WEBRTC_INT16_MAX)
+ return WEBRTC_INT16_MAX;
+ return WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(G722EncoderState *s, int band, int d)
+{
+ int wd1;
+ int wd2;
+ int wd3;
+ int i;
+
+ /* Block 4, RECONS */
+ s->band[band].d[0] = d;
+ s->band[band].r[0] = saturate(s->band[band].s + d);
+
+ /* Block 4, PARREC */
+ s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+ /* Block 4, UPPOL2 */
+ for (i = 0; i < 3; i++)
+ s->band[band].sg[i] = s->band[band].p[i] >> 15;
+ wd1 = saturate(s->band[band].a[1] << 2);
+
+ wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
+ if (wd2 > 32767)
+ wd2 = 32767;
+ wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128);
+ wd3 += (s->band[band].a[2]*32512) >> 15;
+ if (wd3 > 12288)
+ wd3 = 12288;
+ else if (wd3 < -12288)
+ wd3 = -12288;
+ s->band[band].ap[2] = wd3;
+
+ /* Block 4, UPPOL1 */
+ s->band[band].sg[0] = s->band[band].p[0] >> 15;
+ s->band[band].sg[1] = s->band[band].p[1] >> 15;
+ wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
+ wd2 = (s->band[band].a[1]*32640) >> 15;
+
+ s->band[band].ap[1] = saturate(wd1 + wd2);
+ wd3 = saturate(15360 - s->band[band].ap[2]);
+ if (s->band[band].ap[1] > wd3)
+ s->band[band].ap[1] = wd3;
+ else if (s->band[band].ap[1] < -wd3)
+ s->band[band].ap[1] = -wd3;
+
+ /* Block 4, UPZERO */
+ wd1 = (d == 0) ? 0 : 128;
+ s->band[band].sg[0] = d >> 15;
+ for (i = 1; i < 7; i++)
+ {
+ s->band[band].sg[i] = s->band[band].d[i] >> 15;
+ wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
+ wd3 = (s->band[band].b[i]*32640) >> 15;
+ s->band[band].bp[i] = saturate(wd2 + wd3);
+ }
+
+ /* Block 4, DELAYA */
+ for (i = 6; i > 0; i--)
+ {
+ s->band[band].d[i] = s->band[band].d[i - 1];
+ s->band[band].b[i] = s->band[band].bp[i];
+ }
+
+ for (i = 2; i > 0; i--)
+ {
+ s->band[band].r[i] = s->band[band].r[i - 1];
+ s->band[band].p[i] = s->band[band].p[i - 1];
+ s->band[band].a[i] = s->band[band].ap[i];
+ }
+
+ /* Block 4, FILTEP */
+ wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+ wd1 = (s->band[band].a[1]*wd1) >> 15;
+ wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+ wd2 = (s->band[band].a[2]*wd2) >> 15;
+ s->band[band].sp = saturate(wd1 + wd2);
+
+ /* Block 4, FILTEZ */
+ s->band[band].sz = 0;
+ for (i = 6; i > 0; i--)
+ {
+ wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+ s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+ }
+ s->band[band].sz = saturate(s->band[band].sz);
+
+ /* Block 4, PREDIC */
+ s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+ int rate,
+ int options) {
+ if (s == NULL)
+ {
+ if ((s = (G722EncoderState *) malloc(sizeof(*s))) == NULL)
+ return NULL;
+ }
+ memset(s, 0, sizeof(*s));
+ if (rate == 48000)
+ s->bits_per_sample = 6;
+ else if (rate == 56000)
+ s->bits_per_sample = 7;
+ else
+ s->bits_per_sample = 8;
+ if ((options & G722_SAMPLE_RATE_8000))
+ s->eight_k = TRUE;
+ if ((options & G722_PACKED) && s->bits_per_sample != 8)
+ s->packed = TRUE;
+ else
+ s->packed = FALSE;
+ s->band[0].det = 32;
+ s->band[1].det = 8;
+ return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_encode_release(G722EncoderState *s)
+{
+ free(s);
+ return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* WebRtc, tlegrand:
+ * Only define the following if bit-exactness with reference implementation
+ * is needed. Will only have any effect if input signal is saturated.
+ */
+//#define RUN_LIKE_REFERENCE_G722
+#ifdef RUN_LIKE_REFERENCE_G722
+int16_t limitValues (int16_t rl)
+{
+
+ int16_t yl;
+
+ yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
+
+ return (yl);
+}
+#endif
+
+size_t WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
+ const int16_t amp[], size_t len)
+{
+ static const int q6[32] =
+ {
+ 0, 35, 72, 110, 150, 190, 233, 276,
+ 323, 370, 422, 473, 530, 587, 650, 714,
+ 786, 858, 940, 1023, 1121, 1219, 1339, 1458,
+ 1612, 1765, 1980, 2195, 2557, 2919, 0, 0
+ };
+ static const int iln[32] =
+ {
+ 0, 63, 62, 31, 30, 29, 28, 27,
+ 26, 25, 24, 23, 22, 21, 20, 19,
+ 18, 17, 16, 15, 14, 13, 12, 11,
+ 10, 9, 8, 7, 6, 5, 4, 0
+ };
+ static const int ilp[32] =
+ {
+ 0, 61, 60, 59, 58, 57, 56, 55,
+ 54, 53, 52, 51, 50, 49, 48, 47,
+ 46, 45, 44, 43, 42, 41, 40, 39,
+ 38, 37, 36, 35, 34, 33, 32, 0
+ };
+ static const int wl[8] =
+ {
+ -60, -30, 58, 172, 334, 538, 1198, 3042
+ };
+ static const int rl42[16] =
+ {
+ 0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
+ };
+ static const int ilb[32] =
+ {
+ 2048, 2093, 2139, 2186, 2233, 2282, 2332,
+ 2383, 2435, 2489, 2543, 2599, 2656, 2714,
+ 2774, 2834, 2896, 2960, 3025, 3091, 3158,
+ 3228, 3298, 3371, 3444, 3520, 3597, 3676,
+ 3756, 3838, 3922, 4008
+ };
+ static const int qm4[16] =
+ {
+ 0, -20456, -12896, -8968,
+ -6288, -4240, -2584, -1200,
+ 20456, 12896, 8968, 6288,
+ 4240, 2584, 1200, 0
+ };
+ static const int qm2[4] =
+ {
+ -7408, -1616, 7408, 1616
+ };
+ static const int qmf_coeffs[12] =
+ {
+ 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
+ };
+ static const int ihn[3] = {0, 1, 0};
+ static const int ihp[3] = {0, 3, 2};
+ static const int wh[3] = {0, -214, 798};
+ static const int rh2[4] = {2, 1, 2, 1};
+
+ int dlow;
+ int dhigh;
+ int el;
+ int wd;
+ int wd1;
+ int ril;
+ int wd2;
+ int il4;
+ int ih2;
+ int wd3;
+ int eh;
+ int mih;
+ int i;
+ size_t j;
+ /* Low and high band PCM from the QMF */
+ int xlow;
+ int xhigh;
+ size_t g722_bytes;
+ /* Even and odd tap accumulators */
+ int sumeven;
+ int sumodd;
+ int ihigh;
+ int ilow;
+ int code;
+
+ g722_bytes = 0;
+ xhigh = 0;
+ for (j = 0; j < len; )
+ {
+ if (s->itu_test_mode)
+ {
+ xlow =
+ xhigh = amp[j++] >> 1;
+ }
+ else
+ {
+ if (s->eight_k)
+ {
+ /* We shift by 1 to allow for the 15 bit input to the G.722 algorithm. */
+ xlow = amp[j++] >> 1;
+ }
+ else
+ {
+ /* Apply the transmit QMF */
+ /* Shuffle the buffer down */
+ for (i = 0; i < 22; i++)
+ s->x[i] = s->x[i + 2];
+ s->x[22] = amp[j++];
+ s->x[23] = amp[j++];
+
+ /* Discard every other QMF output */
+ sumeven = 0;
+ sumodd = 0;
+ for (i = 0; i < 12; i++)
+ {
+ sumodd += s->x[2*i]*qmf_coeffs[i];
+ sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
+ }
+ /* We shift by 12 to allow for the QMF filters (DC gain = 4096), plus 1
+ to allow for us summing two filters, plus 1 to allow for the 15 bit
+ input to the G.722 algorithm. */
+ xlow = (sumeven + sumodd) >> 14;
+ xhigh = (sumeven - sumodd) >> 14;
+
+#ifdef RUN_LIKE_REFERENCE_G722
+ /* The following lines are only used to verify bit-exactness
+ * with reference implementation of G.722. Higher precision
+ * is achieved without limiting the values.
+ */
+ xlow = limitValues(xlow);
+ xhigh = limitValues(xhigh);
+#endif
+ }
+ }
+ /* Block 1L, SUBTRA */
+ el = saturate(xlow - s->band[0].s);
+
+ /* Block 1L, QUANTL */
+ wd = (el >= 0) ? el : -(el + 1);
+
+ for (i = 1; i < 30; i++)
+ {
+ wd1 = (q6[i]*s->band[0].det) >> 12;
+ if (wd < wd1)
+ break;
+ }
+ ilow = (el < 0) ? iln[i] : ilp[i];
+
+ /* Block 2L, INVQAL */
+ ril = ilow >> 2;
+ wd2 = qm4[ril];
+ dlow = (s->band[0].det*wd2) >> 15;
+
+ /* Block 3L, LOGSCL */
+ il4 = rl42[ril];
+ wd = (s->band[0].nb*127) >> 7;
+ s->band[0].nb = wd + wl[il4];
+ if (s->band[0].nb < 0)
+ s->band[0].nb = 0;
+ else if (s->band[0].nb > 18432)
+ s->band[0].nb = 18432;
+
+ /* Block 3L, SCALEL */
+ wd1 = (s->band[0].nb >> 6) & 31;
+ wd2 = 8 - (s->band[0].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[0].det = wd3 << 2;
+
+ block4(s, 0, dlow);
+
+ if (s->eight_k)
+ {
+ /* Just leave the high bits as zero */
+ code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
+ }
+ else
+ {
+ /* Block 1H, SUBTRA */
+ eh = saturate(xhigh - s->band[1].s);
+
+ /* Block 1H, QUANTH */
+ wd = (eh >= 0) ? eh : -(eh + 1);
+ wd1 = (564*s->band[1].det) >> 12;
+ mih = (wd >= wd1) ? 2 : 1;
+ ihigh = (eh < 0) ? ihn[mih] : ihp[mih];
+
+ /* Block 2H, INVQAH */
+ wd2 = qm2[ihigh];
+ dhigh = (s->band[1].det*wd2) >> 15;
+
+ /* Block 3H, LOGSCH */
+ ih2 = rh2[ihigh];
+ wd = (s->band[1].nb*127) >> 7;
+ s->band[1].nb = wd + wh[ih2];
+ if (s->band[1].nb < 0)
+ s->band[1].nb = 0;
+ else if (s->band[1].nb > 22528)
+ s->band[1].nb = 22528;
+
+ /* Block 3H, SCALEH */
+ wd1 = (s->band[1].nb >> 6) & 31;
+ wd2 = 10 - (s->band[1].nb >> 11);
+ wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
+ s->band[1].det = wd3 << 2;
+
+ block4(s, 1, dhigh);
+ code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
+ }
+
+ if (s->packed)
+ {
+ /* Pack the code bits */
+ s->out_buffer |= (code << s->out_bits);
+ s->out_bits += s->bits_per_sample;
+ if (s->out_bits >= 8)
+ {
+ g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
+ s->out_bits -= 8;
+ s->out_buffer >>= 8;
+ }
+ }
+ else
+ {
+ g722_data[g722_bytes++] = (uint8_t) code;
+ }
+ }
+ return g722_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_interfacec"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,107 @@
</span><ins>+/*
+ * Copyright (c) 2011 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 <stdlib.h>
+#include <string.h>
+#include "g722_enc_dec.h"
+#include "g722_interface.h"
+#include "webrtc/typedefs.h"
+
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+ *G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
+ if (*G722enc_inst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+{
+ // Create and/or reset the G.722 encoder
+ // Bitrate 64 kbps and wideband mode (2)
+ G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
+ (G722EncoderState*) G722enc_inst, 64000, 2);
+ if (G722enc_inst == NULL) {
+ return -1;
+ } else {
+ return 0;
+ }
+}
+
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+ // Free encoder memory
+ return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
+}
+
+size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+ const int16_t* speechIn,
+ size_t len,
+ uint8_t* encoded)
+{
+ unsigned char *codechar = (unsigned char*) encoded;
+ // Encode the input speech vector
+ return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
+ speechIn, len);
+}
+
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+ *G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
+ if (*G722dec_inst!=NULL) {
+ return(0);
+ } else {
+ return(-1);
+ }
+}
+
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+ // Create and/or reset the G.722 decoder
+ // Bitrate 64 kbps and wideband mode (2)
+ WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
+}
+
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+ // Free encoder memory
+ return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
+}
+
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ const uint8_t *encoded,
+ size_t len,
+ int16_t *decoded,
+ int16_t *speechType)
+{
+ // Decode the G.722 encoder stream
+ *speechType=G722_WEBRTC_SPEECH;
+ return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
+ encoded, len);
+}
+
+int16_t WebRtcG722_Version(char *versionStr, short len)
+{
+ // Get version string
+ char version[30] = "2.0.0\n";
+ if (strlen(version) < (unsigned int)len)
+ {
+ strcpy(versionStr, version);
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722g722_interfaceh"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,182 @@
</span><ins>+/*
+ * Copyright (c) 2011 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_G722_G722_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+
+#include "webrtc/typedefs.h"
+
+/*
+ * Solution to support multiple instances
+ */
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/*
+ * Comfort noise constants
+ */
+
+#define G722_WEBRTC_SPEECH 1
+#define G722_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+/****************************************************************************
+ * WebRtcG722_CreateEncoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ * - G722enc_inst : G722 instance for encoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_EncoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ * - G722enc_inst : G722 instance, i.e. the user that should receive
+ * be initialized
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_FreeEncoder(...)
+ *
+ * Free the memory used for G722 encoder
+ *
+ * Input:
+ * - G722enc_inst : G722 instance for encoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+
+
+
+/****************************************************************************
+ * WebRtcG722_Encode(...)
+ *
+ * This function encodes G722 encoded data.
+ *
+ * Input:
+ * - G722enc_inst : G722 instance, i.e. the user that should encode
+ * a packet
+ * - speechIn : Input speech vector
+ * - len : Samples in speechIn
+ *
+ * Output:
+ * - encoded : The encoded data vector
+ *
+ * Return value : Length (in bytes) of coded data
+ */
+
+size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
+ const int16_t* speechIn,
+ size_t len,
+ uint8_t* encoded);
+
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ * - G722dec_inst : G722 instance for decoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_DecoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ * - inst : G722 instance
+ */
+
+void WebRtcG722_DecoderInit(G722DecInst* inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeDecoder(...)
+ *
+ * Free the memory used for G722 decoder
+ *
+ * Input:
+ * - G722dec_inst : G722 instance for decoder
+ *
+ * Return value : 0 - Ok
+ * -1 - Error
+ */
+
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+
+
+/****************************************************************************
+ * WebRtcG722_Decode(...)
+ *
+ * This function decodes a packet with G729 frame(s). Output speech length
+ * will be a multiple of 80 samples (80*frames/packet).
+ *
+ * Input:
+ * - G722dec_inst : G722 instance, i.e. the user that should decode
+ * a packet
+ * - encoded : Encoded G722 frame(s)
+ * - len : Bytes in encoded vector
+ *
+ * Output:
+ * - decoded : The decoded vector
+ * - speechType : 1 normal, 2 CNG (Since G722 does not have its own
+ * DTX/CNG scheme it should always return 1)
+ *
+ * Return value : Samples in decoded vector
+ */
+
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+ const uint8_t* encoded,
+ size_t len,
+ int16_t *decoded,
+ int16_t *speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+int16_t WebRtcG722_Version(char *versionStr, short len);
+
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesaudio_codingcodecsg722testtestG722cc"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc (0 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc 2017-02-10 19:03:56 UTC (rev 212126)
</span><span class="lines">@@ -0,0 +1,158 @@
</span><ins>+/*
+ * 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.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include "webrtc/typedefs.h"
+
+/* include API */
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722 100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t *data, FILE *inp, size_t length)
+{
+ size_t rlen = fread(data, sizeof(int16_t), length, inp);
+ if (rlen >= length)
+ return false;
+ memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+ return true;
+}
+
+int main(int argc, char* argv[])
+{
+ char inname[60], outbit[40], outname[40];
+ FILE *inp, *outbitp, *outp;
+
+ int framecnt;
+ bool endfile;
+ size_t framelength = 160;
+ G722EncInst *G722enc_inst;
+ G722DecInst *G722dec_inst;
+
+ /* Runtime statistics */
+ double starttime;
+ double runtime = 0;
+ double length_file;
+
+ size_t stream_len = 0;
+ int16_t shortdata[960];
+ int16_t decoded[960];
+ uint8_t streamdata[80 * 6];
+ int16_t speechType[1];
+
+ /* handling wrong input arguments in the command line */
+ if (argc!=5) {
+ printf("\n\nWrong number of arguments or flag values.\n\n");
+
+ printf("\n");
+ printf("Usage:\n\n");
+ printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+ printf("with:\n");
+ printf("framelength : Framelength in samples.\n\n");
+ printf("infile : Normal speech input file\n\n");
+ printf("outbitfile : Bitstream output file\n\n");
+ printf("outspeechfile: Speech output file\n\n");
+ exit(0);
+
+ }
+
+ /* Get frame length */
+ int framelength_int = atoi(argv[1]);
+ if (framelength_int < 0) {
+ printf(" G.722: Invalid framelength %d.\n", framelength_int);
+ exit(1);
+ }
+ framelength = static_cast<size_t>(framelength_int);
+
+ /* Get Input and Output files */
+ sscanf(argv[2], "%s", inname);
+ sscanf(argv[3], "%s", outbit);
+ sscanf(argv[4], "%s", outname);
+
+ if ((inp = fopen(inname,"rb")) == NULL) {
+ printf(" G.722: Cannot read file %s.\n", inname);
+ exit(1);
+ }
+ if ((outbitp = fopen(outbit,"wb")) == NULL) {
+ printf(" G.722: Cannot write file %s.\n", outbit);
+ exit(1);
+ }
+ if ((outp = fopen(outname,"wb")) == NULL) {
+ printf(" G.722: Cannot write file %s.\n", outname);
+ exit(1);
+ }
+ printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit, outname);
+
+ /* Create and init */
+ WebRtcG722_CreateEncoder((G722EncInst **)&G722enc_inst);
+ WebRtcG722_CreateDecoder((G722DecInst **)&G722dec_inst);
+ WebRtcG722_EncoderInit((G722EncInst *)G722enc_inst);
+ WebRtcG722_DecoderInit((G722DecInst *)G722dec_inst);
+
+
+ /* Initialize encoder and decoder */
+ framecnt = 0;
+ endfile = false;
+ while (!endfile) {
+ framecnt++;
+
+ /* Read speech block */
+ endfile = readframe(shortdata, inp, framelength);
+
+ /* Start clock before call to encoder and decoder */
+ starttime = clock()/(double)CLOCKS_PER_SEC_G722;
+
+ /* G.722 encoding + decoding */
+ stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
+ WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
+ speechType);
+
+ /* Stop clock after call to encoder and decoder */
+ runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
+
+ /* Write coded bits to file */
+ if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
+ stream_len / 2) {
+ return -1;
+ }
+ /* Write coded speech to file */
+ if (fwrite(decoded, sizeof(short), framelength, outp) !=
+ framelength) {
+ return -1;
+ }
+ }
+
+ WebRtcG722_FreeEncoder((G722EncInst *)G722enc_inst);
+ WebRtcG722_FreeDecoder((G722DecInst *)G722dec_inst);
+
+ length_file = ((double)framecnt*(double)framelength/16000);
+ printf("\n\nLength of speech file: %.1f s\n", length_file);
+ printf("Time to run G.722: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
+ printf("---------------------END----------------------\n");
+
+ fclose(inp);
+ fclose(outbitp);
+ fclose(outp);
+
+ return 0;
+}
</ins></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtclibwebrtcxcodeprojprojectpbxproj"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj (212125 => 212126)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj 2017-02-10 18:56:07 UTC (rev 212125)
+++ trunk/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj 2017-02-10 19:03:56 UTC (rev 212126)
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</ins><span class="cx"> 5C63F9FB1E4174F6002CA531 /* portallocator.h in Headers */,
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</span><span class="cx"> 5CDD8A461E43BFB300621E92 /* post_decode_vad.h in Headers */,
</span><span class="lines">@@ -10107,11 +10177,14 @@
</span><span class="cx"> 5CDD8BDA1E43C2B500621E92 /* refiner.h in Headers */,
</span><span class="cx"> 5C63FA021E4174F6002CA531 /* relayport.h in Headers */,
</span><span class="cx"> 5C63FA051E4174F6002CA531 /* relayserver.h in Headers */,
</span><ins>+ 4140B8341E4E3396007409E6 /* audio_encoder_g722.h in Headers */,
+ 4140B8211E4E3383007409E6 /* audio_encoder_pcm.h in Headers */,
</ins><span class="cx"> 5CDD89621E43BF3A00621E92 /* remb.h in Headers */,
</span><span class="cx"> 5CDD8FDC1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.h in Headers */,
</span><span class="cx"> 5CDD8FDE1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.h in Headers */,
</span><span class="cx"> 5CDD8FE21E43CD6600621E92 /* remote_estimator_proxy.h in Headers */,
</span><span class="cx"> 5C63F9631E41737B002CA531 /* remoteaudiosource.h in Headers */,
</span><ins>+ 4140B8321E4E3396007409E6 /* audio_decoder_g722.h in Headers */,
</ins><span class="cx"> 5CDD84BC1E43AF1300621E92 /* render_queue_item_verifier.h in Headers */,
</span><span class="cx"> 5CDD85191E43B1EA00621E92 /* rent_a_codec.h in Headers */,
</span><span class="cx"> 5CDD89651E43BF3A00621E92 /* report_block.h in Headers */,
</span><span class="lines">@@ -10137,6 +10210,7 @@
</span><span class="cx"> 5C4B4BC11E431C90002651C8 /* RTCAudioSource+Private.h in Headers */,
</span><span class="cx"> 5C4B4BC31E431C90002651C8 /* RTCAudioTrack+Private.h in Headers */,
</span><span class="cx"> 5C4B4BC51E431C90002651C8 /* RTCAVFoundationVideoSource+Private.h in Headers */,
</span><ins>+ 4140B8231E4E3383007409E6 /* audio_decoder_pcm.h in Headers */,
</ins><span class="cx"> 5C63F8651E416288002CA531 /* rtccertificate.h in Headers */,
</span><span class="cx"> 5C63F8671E416288002CA531 /* rtccertificategenerator.h in Headers */,
</span><span class="cx"> 5C4B4BC81E431C90002651C8 /* RTCConfiguration+Private.h in Headers */,
</span><span class="lines">@@ -10300,6 +10374,7 @@
</span><span class="cx"> 5C4B48F51E42C1E3002651C8 /* streamparams.h in Headers */,
</span><span class="cx"> 5C63F89B1E416288002CA531 /* stringencode.h in Headers */,
</span><span class="cx"> 5C63F89D1E416288002CA531 /* stringutils.h in Headers */,
</span><ins>+ 4140B8271E4E3383007409E6 /* g711.h in Headers */,
</ins><span class="cx"> 5CDD87DA1E43BC0600621E92 /* structs.h in Headers */,
</span><span class="cx"> 5C63FA0C1E4174F6002CA531 /* stun.h in Headers */,
</span><span class="cx"> 5C63FA0F1E4174F6002CA531 /* stunport.h in Headers */,
</span><span class="lines">@@ -11458,6 +11533,7 @@
</span><span class="cx"> 5CDD86441E43B8B500621E92 /* complex_bit_reverse.c in Sources */,
</span><span class="cx"> 5CDD86471E43B8B500621E92 /* complex_fft.c in Sources */,
</span><span class="cx"> 5CDD89441E43BF3A00621E92 /* compound_packet.cc in Sources */,
</span><ins>+ 4140B8371E4E3396007409E6 /* g722_encode.c in Sources */,
</ins><span class="cx"> 5CDD8FEB1E43CDCA00621E92 /* config.cc in Sources */,
</span><span class="cx"> 5C6CDD4D1E413598009754E3 /* config.cc in Sources */,
</span><span class="cx"> 5CDD8FB21E43CD0700621E92 /* congestion_controller.cc in Sources */,
</span><span class="lines">@@ -11550,6 +11626,7 @@
</span><span class="cx"> 5C4B4CD41E4320A9002651C8 /* file_impl.cc in Sources */,
</span><span class="cx"> 5CDD84DF1E43B04D00621E92 /* file_player.cc in Sources */,
</span><span class="cx"> 5C63F7E31E416288002CA531 /* file_posix.cc in Sources */,
</span><ins>+ 4140B8311E4E3396007409E6 /* audio_decoder_g722.cc in Sources */,
</ins><span class="cx"> 5CDD84E01E43B04D00621E92 /* file_recorder.cc in Sources */,
</span><span class="cx"> 5CDD8C8B1E43C66000621E92 /* file_utils.cc in Sources */,
</span><span class="cx"> 5C63F7E61E416288002CA531 /* filerotatingstream.cc in Sources */,
</span><span class="lines">@@ -11583,6 +11660,7 @@
</span><span class="cx"> 5CDD84B21E43AF1300621E92 /* gain_control_for_experimental_agc.cc in Sources */,
</span><span class="cx"> 5CDD84B41E43AF1300621E92 /* gain_control_impl.cc in Sources */,
</span><span class="cx"> 5CDD8BA41E43C2B500621E92 /* gain_dequant.c in Sources */,
</span><ins>+ 4140B8331E4E3396007409E6 /* audio_encoder_g722.cc in Sources */,
</ins><span class="cx"> 5CDD8BA61E43C2B500621E92 /* gain_quant.c in Sources */,
</span><span class="cx"> 5CDD90221E43CEDE00621E92 /* gain_selector.cc in Sources */,
</span><span class="cx"> 5CDD83D51E439A6F00621E92 /* generic_decoder.cc in Sources */,
</span><span class="lines">@@ -11596,6 +11674,7 @@
</span><span class="cx"> 5CDD85581E43B42B00621E92 /* h264.cc in Sources */,
</span><span class="cx"> 5CDD83541E43257200621E92 /* h264_bitstream_parser.cc in Sources */,
</span><span class="cx"> 5CDD83561E43257200621E92 /* h264_common.cc in Sources */,
</span><ins>+ 4140B8241E4E3383007409E6 /* g711_interface.c in Sources */,
</ins><span class="cx"> 5CDD85501E43B42B00621E92 /* h264_objc.mm in Sources */,
</span><span class="cx"> 5CDD83D91E439A6F00621E92 /* h264_sps_pps_tracker.cc in Sources */,
</span><span class="cx"> 5C4B4BB71E431C90002651C8 /* h264_video_toolbox_decoder.cc in Sources */,
</span><span class="lines">@@ -11994,6 +12073,7 @@
</span><span class="cx"> 5CDD85BC1E43B5C000621E92 /* stats_counter.cc in Sources */,
</span><span class="cx"> 5C63F9701E41737B002CA531 /* statscollector.cc in Sources */,
</span><span class="cx"> 5C63F9721E41737B002CA531 /* statstypes.cc in Sources */,
</span><ins>+ 4140B8381E4E3396007409E6 /* g722_interface.c in Sources */,
</ins><span class="cx"> 5C63F8981E416288002CA531 /* stream.cc in Sources */,
</span><span class="cx"> 5CDD85BF1E43B5C000621E92 /* stream_synchronization.cc in Sources */,
</span><span class="cx"> 5C4B48F41E42C1E3002651C8 /* streamparams.cc in Sources */,
</span><span class="lines">@@ -12019,6 +12099,8 @@
</span><span class="cx"> 5CDD84C31E43AF1300621E92 /* three_band_filter_bank.cc in Sources */,
</span><span class="cx"> 5CDD8A571E43BFB300621E92 /* tick_timer.cc in Sources */,
</span><span class="cx"> 5CDD87F71E43BDED00621E92 /* time_scheduler.cc in Sources */,
</span><ins>+ 4140B8261E4E3383007409E6 /* g711.c in Sources */,
+ 4140B8221E4E3383007409E6 /* audio_decoder_pcm.cc in Sources */,
</ins><span class="cx"> 5CDD8A5A1E43BFB300621E92 /* time_stretch.cc in Sources */,
</span><span class="cx"> 5CDD88DA1E43BE3D00621E92 /* time_util.cc in Sources */,
</span><span class="cx"> 5C4B4CE91E4320A9002651C8 /* timestamp_extrapolator.cc in Sources */,
</span><span class="lines">@@ -12064,6 +12146,7 @@
</span><span class="cx"> 5CDD86E31E43BA2800621E92 /* vad.cc in Sources */,
</span><span class="cx"> 5CDD873A1E43BABE00621E92 /* vad_audio_proc.cc in Sources */,
</span><span class="cx"> 5CDD873D1E43BABE00621E92 /* vad_circular_buffer.cc in Sources */,
</span><ins>+ 4140B8351E4E3396007409E6 /* g722_decode.c in Sources */,
</ins><span class="cx"> 5CDD86D61E43BA2800621E92 /* vad_core.c in Sources */,
</span><span class="cx"> 5CDD86D91E43BA2800621E92 /* vad_filterbank.c in Sources */,
</span><span class="cx"> 5CDD86DC1E43BA2800621E92 /* vad_gmm.c in Sources */,
</span><span class="lines">@@ -12103,6 +12186,7 @@
</span><span class="cx"> 5CDD847B1E43AE2900621E92 /* voe_network_impl.cc in Sources */,
</span><span class="cx"> 5CDD847D1E43AE2900621E92 /* voe_rtp_rtcp_impl.cc in Sources */,
</span><span class="cx"> 5CDD847F1E43AE2900621E92 /* voe_video_sync_impl.cc in Sources */,
</span><ins>+ 4140B8201E4E3383007409E6 /* audio_encoder_pcm.cc in Sources */,
</ins><span class="cx"> 5CDD84811E43AE2900621E92 /* voe_volume_control_impl.cc in Sources */,
</span><span class="cx"> 5CDD87401E43BABE00621E92 /* voice_activity_detector.cc in Sources */,
</span><span class="cx"> 5CDD84C71E43AF1300621E92 /* voice_detection_impl.cc in Sources */,
</span></span></pre>
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