Index: webrtc/modules/audio_device/test/audio_device_test_api.cc |
diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc |
index c09b88d5d7925ec44d42ba8e5f728c87037a2cde..88b377ea41e01f55ac8950e1a9f1e24936c0949d 100644 |
--- a/webrtc/modules/audio_device/test/audio_device_test_api.cc |
+++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc |
@@ -85,7 +85,7 @@ class AudioTransportAPI: public AudioTransport { |
int32_t RecordedDataIsAvailable(const void* audioSamples, |
const size_t nSamples, |
const size_t nBytesPerSample, |
- const uint8_t nChannels, |
+ const size_t nChannels, |
const uint32_t sampleRate, |
const uint32_t totalDelay, |
const int32_t clockSkew, |
@@ -110,7 +110,7 @@ class AudioTransportAPI: public AudioTransport { |
int32_t NeedMorePlayData(const size_t nSamples, |
const size_t nBytesPerSample, |
- const uint8_t nChannels, |
+ const size_t nChannels, |
const uint32_t sampleRate, |
void* audioSamples, |
size_t& nSamplesOut, |
@@ -128,29 +128,6 @@ class AudioTransportAPI: public AudioTransport { |
return 0; |
} |
- int OnDataAvailable(const int voe_channels[], |
- int number_of_voe_channels, |
- const int16_t* audio_data, |
- int sample_rate, |
- int number_of_channels, |
- size_t number_of_frames, |
- int audio_delay_milliseconds, |
- int current_volume, |
- bool key_pressed, |
- bool need_audio_processing) override { |
- return 0; |
- } |
- |
- void PushCaptureData(int voe_channel, const void* audio_data, |
- int bits_per_sample, int sample_rate, |
- int number_of_channels, |
- size_t number_of_frames) override {} |
- |
- void PullRenderData(int bits_per_sample, int sample_rate, |
- int number_of_channels, size_t number_of_frames, |
- void* audio_data, |
- int64_t* elapsed_time_ms, |
- int64_t* ntp_time_ms) override {} |
private: |
uint32_t rec_count_; |
uint32_t play_count_; |