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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/audio_processing/include/mock_audio_processing.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/modules/audio_processing/include/mock_audio_processing.h')
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diff --git a/webrtc/modules/audio_processing/include/mock_audio_processing.h b/webrtc/modules/audio_processing/include/mock_audio_processing.h
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+++ b/webrtc/modules/audio_processing/include/mock_audio_processing.h
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+/*
+ * Copyright (c) 2013 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_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class MockEchoCancellation : public EchoCancellation {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_METHOD1(enable_drift_compensation,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_drift_compensation_enabled,
+ bool());
+ MOCK_METHOD1(set_stream_drift_samples,
+ void(int drift));
+ MOCK_CONST_METHOD0(stream_drift_samples,
+ int());
+ MOCK_METHOD1(set_suppression_level,
+ int(SuppressionLevel level));
+ MOCK_CONST_METHOD0(suppression_level,
+ SuppressionLevel());
+ MOCK_CONST_METHOD0(stream_has_echo,
+ bool());
+ MOCK_METHOD1(enable_metrics,
+ int(bool enable));
+ MOCK_CONST_METHOD0(are_metrics_enabled,
+ bool());
+ MOCK_METHOD1(GetMetrics,
+ int(Metrics* metrics));
+ MOCK_METHOD1(enable_delay_logging,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_delay_logging_enabled,
+ bool());
+ MOCK_METHOD2(GetDelayMetrics,
+ int(int* median, int* std));
+ MOCK_METHOD3(GetDelayMetrics,
+ int(int* median, int* std, float* fraction_poor_delays));
+ MOCK_CONST_METHOD0(aec_core,
+ struct AecCore*());
+};
+
+class MockEchoControlMobile : public EchoControlMobile {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_METHOD1(set_routing_mode,
+ int(RoutingMode mode));
+ MOCK_CONST_METHOD0(routing_mode,
+ RoutingMode());
+ MOCK_METHOD1(enable_comfort_noise,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_comfort_noise_enabled,
+ bool());
+ MOCK_METHOD2(SetEchoPath,
+ int(const void* echo_path, size_t size_bytes));
+ MOCK_CONST_METHOD2(GetEchoPath,
+ int(void* echo_path, size_t size_bytes));
+};
+
+class MockGainControl : public GainControl {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_METHOD1(set_stream_analog_level,
+ int(int level));
+ MOCK_METHOD0(stream_analog_level,
+ int());
+ MOCK_METHOD1(set_mode,
+ int(Mode mode));
+ MOCK_CONST_METHOD0(mode,
+ Mode());
+ MOCK_METHOD1(set_target_level_dbfs,
+ int(int level));
+ MOCK_CONST_METHOD0(target_level_dbfs,
+ int());
+ MOCK_METHOD1(set_compression_gain_db,
+ int(int gain));
+ MOCK_CONST_METHOD0(compression_gain_db,
+ int());
+ MOCK_METHOD1(enable_limiter,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_limiter_enabled,
+ bool());
+ MOCK_METHOD2(set_analog_level_limits,
+ int(int minimum, int maximum));
+ MOCK_CONST_METHOD0(analog_level_minimum,
+ int());
+ MOCK_CONST_METHOD0(analog_level_maximum,
+ int());
+ MOCK_CONST_METHOD0(stream_is_saturated,
+ bool());
+};
+
+class MockHighPassFilter : public HighPassFilter {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+};
+
+class MockLevelEstimator : public LevelEstimator {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_METHOD0(RMS,
+ int());
+};
+
+class MockNoiseSuppression : public NoiseSuppression {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_METHOD1(set_level,
+ int(Level level));
+ MOCK_CONST_METHOD0(level,
+ Level());
+ MOCK_CONST_METHOD0(speech_probability,
+ float());
+};
+
+class MockVoiceDetection : public VoiceDetection {
+ public:
+ MOCK_METHOD1(Enable,
+ int(bool enable));
+ MOCK_CONST_METHOD0(is_enabled,
+ bool());
+ MOCK_CONST_METHOD0(stream_has_voice,
+ bool());
+ MOCK_METHOD1(set_stream_has_voice,
+ int(bool has_voice));
+ MOCK_METHOD1(set_likelihood,
+ int(Likelihood likelihood));
+ MOCK_CONST_METHOD0(likelihood,
+ Likelihood());
+ MOCK_METHOD1(set_frame_size_ms,
+ int(int size));
+ MOCK_CONST_METHOD0(frame_size_ms,
+ int());
+};
+
+class MockAudioProcessing : public AudioProcessing {
+ public:
+ MockAudioProcessing()
+ : echo_cancellation_(new MockEchoCancellation),
+ echo_control_mobile_(new MockEchoControlMobile),
+ gain_control_(new MockGainControl),
+ high_pass_filter_(new MockHighPassFilter),
+ level_estimator_(new MockLevelEstimator),
+ noise_suppression_(new MockNoiseSuppression),
+ voice_detection_(new MockVoiceDetection) {
+ }
+
+ virtual ~MockAudioProcessing() {
+ }
+
+ MOCK_METHOD0(Initialize,
+ int());
+ MOCK_METHOD6(Initialize,
+ int(int sample_rate_hz,
+ int output_sample_rate_hz,
+ int reverse_sample_rate_hz,
+ ChannelLayout input_layout,
+ ChannelLayout output_layout,
+ ChannelLayout reverse_layout));
+ MOCK_METHOD1(Initialize,
+ int(const ProcessingConfig& processing_config));
+ MOCK_METHOD1(SetExtraOptions,
+ void(const Config& config));
+ MOCK_METHOD1(set_sample_rate_hz,
+ int(int rate));
+ MOCK_CONST_METHOD0(input_sample_rate_hz,
+ int());
+ MOCK_CONST_METHOD0(sample_rate_hz,
+ int());
+ MOCK_CONST_METHOD0(proc_sample_rate_hz,
+ int());
+ MOCK_CONST_METHOD0(proc_split_sample_rate_hz,
+ int());
+ MOCK_CONST_METHOD0(num_input_channels,
+ int());
+ MOCK_CONST_METHOD0(num_output_channels,
+ int());
+ MOCK_CONST_METHOD0(num_reverse_channels,
+ int());
+ MOCK_METHOD1(set_output_will_be_muted,
+ void(bool muted));
+ MOCK_CONST_METHOD0(output_will_be_muted,
+ bool());
+ MOCK_METHOD1(ProcessStream,
+ int(AudioFrame* frame));
+ MOCK_METHOD7(ProcessStream,
+ int(const float* const* src,
+ size_t samples_per_channel,
+ int input_sample_rate_hz,
+ ChannelLayout input_layout,
+ int output_sample_rate_hz,
+ ChannelLayout output_layout,
+ float* const* dest));
+ MOCK_METHOD4(ProcessStream,
+ int(const float* const* src,
+ const StreamConfig& input_config,
+ const StreamConfig& output_config,
+ float* const* dest));
+ MOCK_METHOD1(AnalyzeReverseStream,
+ int(AudioFrame* frame));
+ MOCK_METHOD1(ProcessReverseStream, int(AudioFrame* frame));
+ MOCK_METHOD4(AnalyzeReverseStream,
+ int(const float* const* data, size_t frames, int sample_rate_hz,
+ ChannelLayout input_layout));
+ MOCK_METHOD4(ProcessReverseStream,
+ int(const float* const* src,
+ const StreamConfig& input_config,
+ const StreamConfig& output_config,
+ float* const* dest));
+ MOCK_METHOD1(set_stream_delay_ms,
+ int(int delay));
+ MOCK_CONST_METHOD0(stream_delay_ms,
+ int());
+ MOCK_CONST_METHOD0(was_stream_delay_set,
+ bool());
+ MOCK_METHOD1(set_stream_key_pressed,
+ void(bool key_pressed));
+ MOCK_CONST_METHOD0(stream_key_pressed,
+ bool());
+ MOCK_METHOD1(set_delay_offset_ms,
+ void(int offset));
+ MOCK_CONST_METHOD0(delay_offset_ms,
+ int());
+ MOCK_METHOD1(StartDebugRecording,
+ int(const char filename[kMaxFilenameSize]));
+ MOCK_METHOD1(StartDebugRecording,
+ int(FILE* handle));
+ MOCK_METHOD0(StopDebugRecording,
+ int());
+ MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
+ virtual MockEchoCancellation* echo_cancellation() const {
+ return echo_cancellation_.get();
+ }
+ virtual MockEchoControlMobile* echo_control_mobile() const {
+ return echo_control_mobile_.get();
+ }
+ virtual MockGainControl* gain_control() const {
+ return gain_control_.get();
+ }
+ virtual MockHighPassFilter* high_pass_filter() const {
+ return high_pass_filter_.get();
+ }
+ virtual MockLevelEstimator* level_estimator() const {
+ return level_estimator_.get();
+ }
+ virtual MockNoiseSuppression* noise_suppression() const {
+ return noise_suppression_.get();
+ }
+ virtual MockVoiceDetection* voice_detection() const {
+ return voice_detection_.get();
+ }
+
+ private:
+ rtc::scoped_ptr<MockEchoCancellation> echo_cancellation_;
+ rtc::scoped_ptr<MockEchoControlMobile> echo_control_mobile_;
+ rtc::scoped_ptr<MockGainControl> gain_control_;
+ rtc::scoped_ptr<MockHighPassFilter> high_pass_filter_;
+ rtc::scoped_ptr<MockLevelEstimator> level_estimator_;
+ rtc::scoped_ptr<MockNoiseSuppression> noise_suppression_;
+ rtc::scoped_ptr<MockVoiceDetection> voice_detection_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_