(Auto)update libjingle 71240799-> 71250251
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index a5b6031..49b78bc 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -40,6 +40,9 @@
#include "talk/media/base/voiceprocessor.h"
#include "talk/media/webrtc/fakewebrtccommon.h"
#include "talk/media/webrtc/webrtcvoe.h"
+#ifdef USE_WEBRTC_DEV_BRANCH
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#endif
namespace webrtc {
class ViENetwork;
@@ -70,6 +73,88 @@
} \
} while (0);
+#ifdef USE_WEBRTC_DEV_BRANCH
+class FakeAudioProcessing : public webrtc::AudioProcessing {
+ public:
+ FakeAudioProcessing() : experimental_ns_enabled_(false) {}
+
+ WEBRTC_STUB(Initialize, ())
+ WEBRTC_STUB(Initialize, (
+ int input_sample_rate_hz,
+ int output_sample_rate_hz,
+ int reverse_sample_rate_hz,
+ webrtc::AudioProcessing::ChannelLayout input_layout,
+ webrtc::AudioProcessing::ChannelLayout output_layout,
+ webrtc::AudioProcessing::ChannelLayout reverse_layout));
+
+ WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
+ experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
+ }
+
+ WEBRTC_STUB(set_sample_rate_hz, (int rate));
+ WEBRTC_STUB_CONST(input_sample_rate_hz, ());
+ WEBRTC_STUB_CONST(sample_rate_hz, ());
+ WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
+ WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
+ WEBRTC_STUB_CONST(num_input_channels, ());
+ WEBRTC_STUB_CONST(num_output_channels, ());
+ WEBRTC_STUB_CONST(num_reverse_channels, ());
+ WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
+ WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ());
+ WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
+ WEBRTC_STUB(ProcessStream, (
+ const float* const* src,
+ int samples_per_channel,
+ int input_sample_rate_hz,
+ webrtc::AudioProcessing::ChannelLayout input_layout,
+ int output_sample_rate_hz,
+ webrtc::AudioProcessing::ChannelLayout output_layout,
+ float* const* dest));
+ WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
+ WEBRTC_STUB(AnalyzeReverseStream, (
+ const float* const* data,
+ int samples_per_channel,
+ int sample_rate_hz,
+ webrtc::AudioProcessing::ChannelLayout layout));
+ WEBRTC_STUB(set_stream_delay_ms, (int delay));
+ WEBRTC_STUB_CONST(stream_delay_ms, ());
+ WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
+ WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
+ WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ());
+ WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
+ WEBRTC_STUB_CONST(delay_offset_ms, ());
+ WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
+ WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+ WEBRTC_STUB(StopDebugRecording, ());
+ virtual webrtc::EchoCancellation* echo_cancellation() const OVERRIDE {
+ return NULL;
+ }
+ virtual webrtc::EchoControlMobile* echo_control_mobile() const OVERRIDE {
+ return NULL;
+ }
+ virtual webrtc::GainControl* gain_control() const OVERRIDE { return NULL; }
+ virtual webrtc::HighPassFilter* high_pass_filter() const OVERRIDE {
+ return NULL;
+ }
+ virtual webrtc::LevelEstimator* level_estimator() const OVERRIDE {
+ return NULL;
+ }
+ virtual webrtc::NoiseSuppression* noise_suppression() const OVERRIDE {
+ return NULL;
+ }
+ virtual webrtc::VoiceDetection* voice_detection() const OVERRIDE {
+ return NULL;
+ }
+
+ bool experimental_ns_enabled() {
+ return experimental_ns_enabled_;
+ }
+
+ private:
+ bool experimental_ns_enabled_;
+};
+#endif
+
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
@@ -347,7 +432,11 @@
return 0;
}
virtual webrtc::AudioProcessing* audio_processing() OVERRIDE {
+#ifdef USE_WEBRTC_DEV_BRANCH
+ return &audio_processing_;
+#else
return NULL;
+#endif
}
WEBRTC_FUNC(CreateChannel, ()) {
return AddChannel();
@@ -1197,6 +1286,9 @@
int playout_sample_rate_;
DtmfInfo dtmf_info_;
webrtc::VoEMediaProcess* media_processor_;
+#ifdef USE_WEBRTC_DEV_BRANCH
+ FakeAudioProcessing audio_processing_;
+#endif
};
#undef WEBRTC_CHECK_HEADER_EXTENSION_ID