Chromium Code Reviews| Index: webrtc/modules/audio_processing/test/audio_buffer_tools.cc |
| diff --git a/webrtc/modules/audio_processing/test/audio_buffer_tools.cc b/webrtc/modules/audio_processing/test/audio_buffer_tools.cc |
| index a8cb09ca6a4b7a3da898163b647c8d42e9ce7d5f..976949c36dd5a55ebdf5172c60f242fb7985dafc 100644 |
| --- a/webrtc/modules/audio_processing/test/audio_buffer_tools.cc |
| +++ b/webrtc/modules/audio_processing/test/audio_buffer_tools.cc |
| @@ -10,10 +10,12 @@ |
| #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h" |
| +#include <string.h> |
| + |
| namespace webrtc { |
| namespace test { |
| -void SetupFrame(StreamConfig stream_config, |
| +void SetupFrame(const StreamConfig& stream_config, |
| std::vector<float*>* frame, |
| std::vector<float>* frame_samples) { |
| frame_samples->resize(stream_config.num_channels() * |
| @@ -25,7 +27,7 @@ void SetupFrame(StreamConfig stream_config, |
| } |
| void CopyVectorToAudioBuffer(const StreamConfig& stream_config, |
| - const std::vector<float>& source, |
| + rtc::ArrayView<const float> source, |
| AudioBuffer* destination) { |
| std::vector<float*> input; |
| std::vector<float> input_samples; |
| @@ -33,22 +35,20 @@ void CopyVectorToAudioBuffer(const StreamConfig& stream_config, |
| SetupFrame(stream_config, &input, &input_samples); |
| RTC_DCHECK_EQ(input_samples.size(), source.size()); |
| - input_samples = source; |
| + memcpy(input_samples.data(), &source[0], |
|
hlundin-webrtc
2016/03/17 14:14:40
&source[0] -> source.data()
peah-webrtc
2016/03/17 22:26:02
Done.
|
| + input_samples.size() * sizeof(input_samples[0])); |
|
hlundin-webrtc
2016/03/17 14:14:40
Using input_samples.size() and sizeof(input_sample
peah-webrtc
2016/03/17 22:26:02
I agree that it is a bit backward. But it ensures
|
| destination->CopyFrom(&input[0], stream_config); |
| } |
| -std::vector<float> ExtractVectorFromAudioBuffer( |
| - const StreamConfig& stream_config, |
| - AudioBuffer* source) { |
| +void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config, |
| + AudioBuffer* source, |
| + std::vector<float>* destination) { |
| std::vector<float*> output; |
| - std::vector<float> output_samples; |
| - SetupFrame(stream_config, &output, &output_samples); |
| + SetupFrame(stream_config, &output, destination); |
| source->CopyTo(stream_config, &output[0]); |
| - |
| - return output_samples; |
| } |
| } // namespace test |