 Chromium Code Reviews
 Chromium Code Reviews Issue 1783693005:
  Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@voe
    
  
    Issue 1783693005:
  Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@voe| Index: webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc | 
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc | 
| index 663a8a51a88a517ed338bf54aa09762f25954160..8134e76b621b33bc2deb843cce00f27521c64089 100644 | 
| --- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc | 
| +++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc | 
| @@ -35,8 +35,7 @@ class AudioProcessingImplLockTest; | 
| enum class RenderApiImpl { | 
| ProcessReverseStreamImpl1, | 
| ProcessReverseStreamImpl2, | 
| - AnalyzeReverseStreamImpl1, | 
| - AnalyzeReverseStreamImpl2 | 
| + AnalyzeReverseStreamImpl1 | 
| 
hlundin-webrtc
2016/03/11 09:35:50
Does the numeral "1" have any meaning now when the
 
aluebs-webrtc
2016/03/11 13:37:55
Good point. Removed the 1.
 
peah-webrtc
2016/03/11 21:25:50
No, it does not, please remove that as part of thi
 
aluebs-webrtc
2016/03/14 16:42:02
Already did.
 | 
| }; | 
| // Type of the capture thread APM API call to use in the test. | 
| @@ -150,7 +149,7 @@ struct TestConfig { | 
| // Create test config for the first processing API function set. | 
| test_configs.push_back(test_config); | 
| test_config.render_api_function = | 
| - RenderApiImpl::AnalyzeReverseStreamImpl2; | 
| + RenderApiImpl::AnalyzeReverseStreamImpl1; | 
| test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3; | 
| test_configs.push_back(test_config); | 
| } | 
| @@ -172,15 +171,13 @@ struct TestConfig { | 
| const AllowedApiCallCombinations api_calls[] = { | 
| {RenderApiImpl::ProcessReverseStreamImpl1, | 
| CaptureApiImpl::ProcessStreamImpl1}, | 
| - {RenderApiImpl::AnalyzeReverseStreamImpl1, | 
| - CaptureApiImpl::ProcessStreamImpl1}, | 
| {RenderApiImpl::ProcessReverseStreamImpl2, | 
| CaptureApiImpl::ProcessStreamImpl2}, | 
| {RenderApiImpl::ProcessReverseStreamImpl2, | 
| CaptureApiImpl::ProcessStreamImpl3}, | 
| - {RenderApiImpl::AnalyzeReverseStreamImpl2, | 
| + {RenderApiImpl::AnalyzeReverseStreamImpl1, | 
| CaptureApiImpl::ProcessStreamImpl2}, | 
| - {RenderApiImpl::AnalyzeReverseStreamImpl2, | 
| + {RenderApiImpl::AnalyzeReverseStreamImpl1, | 
| CaptureApiImpl::ProcessStreamImpl3}}; | 
| std::vector<TestConfig> out; | 
| for (auto api_call : api_calls) { | 
| @@ -941,8 +938,6 @@ void RenderProcessor::PrepareFrame() { | 
| // Restrict to a common fixed sample rate if the AudioFrame interface is | 
| // used. | 
| if ((test_config_->render_api_function == | 
| - RenderApiImpl::AnalyzeReverseStreamImpl1) || | 
| - (test_config_->render_api_function == | 
| RenderApiImpl::ProcessReverseStreamImpl1) || | 
| (test_config_->aec_type != | 
| AecType::BasicWebRtcAecSettingsWithAecMobile)) { | 
| @@ -1004,9 +999,6 @@ void RenderProcessor::CallApmRenderSide() { | 
| frame_data_.output_stream_config, &frame_data_.output_frame[0]); | 
| break; | 
| case RenderApiImpl::AnalyzeReverseStreamImpl1: | 
| - result = apm_->AnalyzeReverseStream(&frame_data_.frame); | 
| - break; | 
| - case RenderApiImpl::AnalyzeReverseStreamImpl2: | 
| result = apm_->AnalyzeReverseStream( | 
| &frame_data_.input_frame[0], frame_data_.input_samples_per_channel, | 
| frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout); |