(Auto)update libjingle 71107853-> 71115715

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/app/webrtc/test/fakeaudiocapturemodule.cc

@@ -730,20 +730,20 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
     uint32_t nSamplesOut = 0;
 #ifdef USE_WEBRTC_DEV_BRANCH
     int64_t elapsed_time_ms = 0;
+#else
+    uint32_t rtp_timestamp = 0;
+#endif
     int64_t ntp_time_ms = 0;
     if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
                                          kNumberOfChannels, kSamplesPerSecond,
                                          rec_buffer_, nSamplesOut,
+#ifdef USE_WEBRTC_DEV_BRANCH
                                          &elapsed_time_ms, &ntp_time_ms) != 0) {
-      ASSERT(false);
-    }
 #else
-    if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
+                                         &rtp_timestamp, &ntp_time_ms) != 0) {
-                                         kNumberOfChannels, kSamplesPerSecond,
+#endif
-                                         rec_buffer_, nSamplesOut) != 0) {
       ASSERT(false);
     }
-#endif
     ASSERT(nSamplesOut == kNumberSamples);
   }
   // The SetBuffer() function ensures that after decoding, the audio buffer

talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc

@@ -84,13 +84,13 @@ class FakeAdmTest : public testing::Test,
                                    const uint8_t nChannels,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
-#ifdef USE_WEBRTC_DEV_BRANCH
                                    uint32_t& nSamplesOut,
+#ifdef USE_WEBRTC_DEV_BRANCH
                                    int64_t* elapsed_time_ms,
-                                   int64_t* ntp_time_ms) {
 #else
-                                   uint32_t& nSamplesOut) {
+                                   uint32_t* rtp_timestamp,
 #endif
+                                   int64_t* ntp_time_ms) {
     ++pull_iterations_;
     const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
     const uint32_t bytes_out = RecordedDataReceived() ?
@@ -99,8 +99,10 @@ class FakeAdmTest : public testing::Test,
     nSamplesOut = bytes_out / nBytesPerSample;
 #ifdef USE_WEBRTC_DEV_BRANCH
     *elapsed_time_ms = 0;
-    *ntp_time_ms = 0;
+#else
+    *rtp_timestamp = 0;
 #endif
+    *ntp_time_ms = 0;
     return 0;
   }
 

talk/media/webrtc/fakewebrtcvoiceengine.h

@@ -789,7 +789,6 @@ class FakeWebRtcVoiceEngine
     channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
     return 0;
   }
-#ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
       unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ -797,7 +796,6 @@ class FakeWebRtcVoiceEngine
     channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1;
    return 0;
   }
-#endif  // USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
       unsigned char id)) {
     WEBRTC_CHECK_CHANNEL(channel);

talk/media/webrtc/webrtcvideoengine.cc

@@ -211,9 +211,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
   virtual int DeliverFrame(unsigned char* buffer,
                            int buffer_size,
                            uint32_t rtp_time_stamp,
-#ifdef USE_WEBRTC_DEV_BRANCH
                            int64_t ntp_time_ms,
-#endif
                            int64_t render_time,
                            void* handle) {
     talk_base::CritScope cs(&crit_);
@@ -226,11 +224,9 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
     int64 elapsed_time_ms =
         (rtp_ts_wraparound_handler_.Unwrap(rtp_time_stamp) -
          capture_start_rtp_time_stamp_) / kVideoCodecClockratekHz;
-#ifdef USE_WEBRTC_DEV_BRANCH
     if (ntp_time_ms > 0) {
       capture_start_ntp_time_ms_ = ntp_time_ms - elapsed_time_ms;
     }
-#endif
     frame_rate_tracker_.Update(1);
     if (renderer_ == NULL) {
       return 0;

talk/media/webrtc/webrtcvoiceengine.cc

@@ -2280,7 +2280,6 @@ bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
 
 bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
     int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
-#ifdef USE_WEBRTC_DEV_BRANCH
   const RtpHeaderExtension* audio_level_extension =
       FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
   if (!SetHeaderExtension(
@@ -2288,7 +2287,6 @@ bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
       audio_level_extension)) {
     return false;
   }
-#endif  // USE_WEBRTC_DEV_BRANCH
 
   const RtpHeaderExtension* send_time_extension =
       FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -1692,11 +1692,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
 TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
   TestSetSendRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
 }
-#ifdef USE_WEBRTC_DEV_BRANCH
 TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(kRtpAudioLevelHeaderExtension);
 }
-#endif  // USE_WEBRTC_DEV_BRANCH
 
 // Test support for absolute send time header extension.
 TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {