Add a gyp variable to fully disable Opus.

Without this, the Chromium gyp run will fail until Opus is added.

This builds and runs Opus on Linux with include_opus=1, and builds with include_opus=0 (including with third_party/opus deleted to simulate the Chromium build).

Review URL: https://webrtc-codereview.appspot.com/902005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2953 4adac7df-926f-26a2-2b94-8c16560cd09d

src/build/common.gypi

@@ -32,10 +32,12 @@
       'webrtc_root%': '<(webrtc_root)',
 
       'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
+      'include_opus%': 0,
     },
     'build_with_chromium%': '<(build_with_chromium)',
     'webrtc_root%': '<(webrtc_root)',
     'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
+    'include_opus%': '<(include_opus)',
 
     # The Chromium common.gypi we use treats all gyp files without
     # chromium_code==1 as third party code. This disables many of the
@@ -62,6 +64,9 @@
     'build_libyuv%': 1,
     'build_libvpx%': 1,
 
+    # Enable to use the Mozilla internal Opus version.
+    'build_with_mozilla%': 0,
+
     'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
 
     'conditions': [
@@ -86,9 +91,6 @@
         # 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,

src/modules/audio_coding/main/source/acm_codec_database.cc

@@ -88,7 +88,7 @@
 #endif
 #ifdef WEBRTC_CODEC_OPUS
     #include "acm_opus.h"
-    #include "opus_interface.h"
+    #include "modules/audio_coding/codecs/opus/interface/opus_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
     #include "acm_speex.h"

src/modules/audio_coding/main/source/acm_opus.cc

@@ -18,7 +18,7 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_OPUS
-#include "opus_interface.h"
+#include "modules/audio_coding/codecs/opus/interface/opus_interface.h"
 #endif
 
 namespace webrtc {

src/modules/audio_coding/main/source/acm_opus.h

@@ -12,9 +12,11 @@
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
 
 #include "acm_generic_codec.h"
-#include "opus_interface.h"
 #include "resampler.h"
 
+struct WebRtcOpusEncInst;
+struct WebRtcOpusDecInst;
+
 namespace webrtc {
 
 class ACMOpus : public ACMGenericCodec {
@@ -48,8 +50,8 @@ class ACMOpus : public ACMGenericCodec {
 
   int16_t SetBitRateSafe(const int32_t rate);
 
-  OpusEncInst* _encoderInstPtr;
-  OpusDecInst* _decoderInstPtr;
+  WebRtcOpusEncInst* _encoderInstPtr;
+  WebRtcOpusDecInst* _decoderInstPtr;
   uint16_t _sampleFreq;
   uint16_t _bitrate;
 };

src/modules/audio_coding/main/source/audio_coding_module.gypi

@@ -15,7 +15,6 @@
       'iLBC',
       'iSAC',
       'iSACFix',
-      'webrtc_opus',
       'PCM16B',
       'NetEq',
       '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
@@ -24,6 +23,12 @@
       '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
     ],
     'audio_coding_defines': [],
+    'conditions': [
+      ['include_opus==1', {
+        'audio_coding_dependencies': ['webrtc_opus',],
+        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+      }],
+    ],
   },
   'targets': [
     {
@@ -38,13 +43,11 @@
       'include_dirs': [
         '../interface',
         '../../../interface',
-        '../../codecs/opus/interface',
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-        '../interface',
-        '../../../interface',
-        '../../codecs/opus/interface',
+          '../interface',
+          '../../../interface',
         ],
       },
       'sources': [

src/modules/modules.gyp

@@ -15,7 +15,6 @@
     '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',
@@ -34,8 +33,10 @@
     'video_render/main/source/video_render.gypi',
     'rtp_rtcp/source/rtp_rtcp.gypi',
   ],
-
   'conditions': [
+    ['include_opus==1', {
+      'includes': ['audio_coding/codecs/opus/opus.gypi',],
+    }],
     ['include_tests==1', {
       'includes': [
         'audio_coding/codecs/isac/isac_test.gypi',