Add mute state field to AudioFrame.

webrtc/modules/audio_processing/audio_processing_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 69 matching lines @@
                cb->num_channels(),
                cb_int.channels());
   for (size_t i = 0; i < cb->num_channels(); ++i) {
     S16ToFloat(cb_int.channels()[i],
                cb->num_frames(),
                cb->channels()[i]);
   }
 }
 
 void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
-  ConvertToFloat(frame.data_, cb);
+  ConvertToFloat(frame.data(), cb);
 }
 
 // Number of channels including the keyboard channel.
 size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
       return 1;
     case AudioProcessing::kMonoAndKeyboard:
     case AudioProcessing::kStereo:
       return 2;
@@ skipping 19 matching lines @@
   for (size_t i = 0; i < samples_per_channel; ++i)
     mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
 }
 
 void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) {
   for (size_t i = 0; i < samples_per_channel; i++) {
     stereo[i * 2 + 1] = stereo[i * 2];
   }
 }
 
-void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) {
+void VerifyChannelsAreEqual(const int16_t* stereo, size_t samples_per_channel) {
   for (size_t i = 0; i < samples_per_channel; i++) {
     EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
   }
 }
 
 void SetFrameTo(AudioFrame* frame, int16_t value) {
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
        ++i) {
-    frame->data_[i] = value;
+    frame_data[i] = value;
   }
 }
 
 void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
   ASSERT_EQ(2u, frame->num_channels_);
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
-    frame->data_[i] = left;
-    frame->data_[i + 1] = right;
+    frame_data[i] = left;
+    frame_data[i + 1] = right;
   }
 }
 
 void ScaleFrame(AudioFrame* frame, float scale) {
+  int16_t* frame_data = frame->mutable_data();
   for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
        ++i) {
-    frame->data_[i] = FloatS16ToS16(frame->data_[i] * scale);
+    frame_data[i] = FloatS16ToS16(frame_data[i] * scale);
   }
 }
 
 bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
   if (frame1.samples_per_channel_ != frame2.samples_per_channel_) {
     return false;
   }
   if (frame1.num_channels_ != frame2.num_channels_) {
     return false;
   }
-  if (memcmp(frame1.data_, frame2.data_,
+  if (memcmp(frame1.data(), frame2.data(),
              frame1.samples_per_channel_ * frame1.num_channels_ *
                  sizeof(int16_t))) {
     return false;
   }
   return true;
 }
 
 void EnableAllAPComponents(AudioProcessing* ap) {
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
   EXPECT_NOERR(ap->echo_control_mobile()->Enable(true));
@@ skipping 22 matching lines @@
 }
 
 // These functions are only used by ApmTest.Process.
 template <class T>
 T AbsValue(T a) {
   return a > 0 ? a: -a;
 }
 
 int16_t MaxAudioFrame(const AudioFrame& frame) {
   const size_t length = frame.samples_per_channel_ * frame.num_channels_;
-  int16_t max_data = AbsValue(frame.data_[0]);
+  const int16_t* frame_data = frame.data();
+  int16_t max_data = AbsValue(frame_data[0]);
   for (size_t i = 1; i < length; i++) {
-    max_data = std::max(max_data, AbsValue(frame.data_[i]));
+    max_data = std::max(max_data, AbsValue(frame_data[i]));
   }
 
   return max_data;
 }
 
 #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 void TestStats(const AudioProcessing::Statistic& test,
                const audioproc::Test::Statistic& reference) {
   EXPECT_EQ(reference.instant(), test.instant);
   EXPECT_EQ(reference.average(), test.average);
@@ skipping 306 matching lines @@
 }
 
 void ApmTest::EnableAllComponents() {
   EnableAllAPComponents(apm_.get());
 }
 
 bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame,
                         ChannelBuffer<float>* cb) {
   // The files always contain stereo audio.
   size_t frame_size = frame->samples_per_channel_ * 2;
-  size_t read_count = fread(frame->data_,
+  size_t read_count = fread(frame->mutable_data(),
                             sizeof(int16_t),
                             frame_size,
                             file);
   if (read_count != frame_size) {
     // Check that the file really ended.
     EXPECT_NE(0, feof(file));
     return false;  // This is expected.
   }
 
   if (frame->num_channels_ == 1) {
-    MixStereoToMono(frame->data_, frame->data_,
+    MixStereoToMono(frame->data(), frame->mutable_data(),
                     frame->samples_per_channel_);
   }
 
   if (cb) {
     ConvertToFloat(*frame, cb);
   }
   return true;
 }
 
 bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame) {
@@ skipping 1037 matching lines @@
          kProcessSampleRates[i],
          kProcessSampleRates[i],
          2,
          2,
          2,
          false);
     int analog_level = 127;
     ASSERT_EQ(0, feof(far_file_));
     ASSERT_EQ(0, feof(near_file_));
     while (ReadFrame(far_file_, revframe_) && ReadFrame(near_file_, frame_)) {
-      CopyLeftToRightChannel(revframe_->data_, revframe_->samples_per_channel_);
+      CopyLeftToRightChannel(revframe_->mutable_data(),
+                             revframe_->samples_per_channel_);
 
       ASSERT_EQ(kNoErr, apm_->ProcessReverseStream(revframe_));
 
-      CopyLeftToRightChannel(frame_->data_, frame_->samples_per_channel_);
+      CopyLeftToRightChannel(frame_->mutable_data(),
+                             frame_->samples_per_channel_);
       frame_->vad_activity_ = AudioFrame::kVadUnknown;
 
       ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0));
       apm_->echo_cancellation()->set_stream_drift_samples(0);
       ASSERT_EQ(kNoErr,
           apm_->gain_control()->set_stream_analog_level(analog_level));
       ASSERT_EQ(kNoErr, apm_->ProcessStream(frame_));
       analog_level = apm_->gain_control()->stream_analog_level();
 
-      VerifyChannelsAreEqual(frame_->data_, frame_->samples_per_channel_);
+      VerifyChannelsAreEqual(frame_->data(), frame_->samples_per_channel_);
     }
     rewind(far_file_);
     rewind(near_file_);
   }
 }
 
 TEST_F(ApmTest, SplittingFilter) {
   // Verify the filter is not active through undistorted audio when:
   // 1. No components are enabled...
   SetFrameTo(frame_, 1000);
@@ skipping 111 matching lines @@
 
       if (msg.channel_size() > 0) {
         ASSERT_EQ(revframe_->num_channels_,
                   static_cast<size_t>(msg.channel_size()));
         for (int i = 0; i < msg.channel_size(); ++i) {
            memcpy(revfloat_cb_->channels()[i],
                   msg.channel(i).data(),
                   msg.channel(i).size());
         }
       } else {
-        memcpy(revframe_->data_, msg.data().data(), msg.data().size());
+        memcpy(revframe_->mutable_data(), msg.data().data(), msg.data().size());
         if (format == kFloatFormat) {
           // We're using an int16 input file; convert to float.
           ConvertToFloat(*revframe_, revfloat_cb_.get());
         }
       }
       AnalyzeReverseStreamChooser(format);
 
     } else if (event_msg.type() == audioproc::Event::STREAM) {
       const audioproc::Stream msg = event_msg.stream();
       // ProcessStream could have changed this for the output frame.
@@ skipping 10 matching lines @@
 
       if (msg.input_channel_size() > 0) {
         ASSERT_EQ(frame_->num_channels_,
                   static_cast<size_t>(msg.input_channel_size()));
         for (int i = 0; i < msg.input_channel_size(); ++i) {
            memcpy(float_cb_->channels()[i],
                   msg.input_channel(i).data(),
                   msg.input_channel(i).size());
         }
       } else {
-        memcpy(frame_->data_, msg.input_data().data(), msg.input_data().size());
+        memcpy(frame_->mutable_data(), msg.input_data().data(),
+               msg.input_data().size());
         if (format == kFloatFormat) {
           // We're using an int16 input file; convert to float.
           ConvertToFloat(*frame_, float_cb_.get());
         }
       }
       ProcessStreamChooser(format);
     }
   }
   EXPECT_NOERR(apm_->StopDebugRecording());
   fclose(in_file);
@@ skipping 188 matching lines @@
           LayoutFromChannels(num_render_channels)));
 
       EXPECT_NOERR(apm_->set_stream_delay_ms(0));
       EXPECT_NOERR(fapm->set_stream_delay_ms(0));
       apm_->echo_cancellation()->set_stream_drift_samples(0);
       fapm->echo_cancellation()->set_stream_drift_samples(0);
       EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(analog_level));
       EXPECT_NOERR(fapm->gain_control()->set_stream_analog_level(analog_level));
 
       EXPECT_NOERR(apm_->ProcessStream(frame_));
-      Deinterleave(frame_->data_, samples_per_channel, num_output_channels,
+      Deinterleave(frame_->data(), samples_per_channel, num_output_channels,
                    output_int16.channels());
 
       EXPECT_NOERR(fapm->ProcessStream(
           float_cb_->channels(),
           samples_per_channel,
           test->sample_rate(),
           LayoutFromChannels(num_input_channels),
           test->sample_rate(),
           LayoutFromChannels(num_output_channels),
           float_cb_->channels()));
@@ skipping 143 matching lines @@
       if (apm_->voice_detection()->stream_has_voice()) {
         has_voice_count++;
         EXPECT_EQ(AudioFrame::kVadActive, frame_->vad_activity_);
       } else {
         EXPECT_EQ(AudioFrame::kVadPassive, frame_->vad_activity_);
       }
 
       ns_speech_prob_average += apm_->noise_suppression()->speech_probability();
 
       size_t frame_size = frame_->samples_per_channel_ * frame_->num_channels_;
-      size_t write_count = fwrite(frame_->data_,
+      size_t write_count = fwrite(frame_->data(),
                                   sizeof(int16_t),
                                   frame_size,
                                   out_file_);
       ASSERT_EQ(frame_size, write_count);
 
       // Reset in case of downmixing.
       frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
       frame_count++;
 
 #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
@@ skipping 709 matching lines @@
   // TODO(peah): Remove the testing for
   // apm->capture_nonlocked_.level_controller_enabled once the value in config_
   // is instead used to activate the level controller.
   EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
   EXPECT_NEAR(kTargetLcPeakLeveldBFS,
               apm->config_.level_controller.initial_peak_level_dbfs,
               std::numeric_limits<float>::epsilon());
 }
 
 }  // namespace webrtc