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APK for opensl loopback.

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BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2212004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4901 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org
2013-10-02 14:58:19 +00:00
parent de74b64184
commit 1fdc51ae2a
17 changed files with 697 additions and 13 deletions
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19
webrtc/modules/audio_device/audio_device_buffer.h
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@ -36,13 +36,13 @@ public:
int32_t InitPlayout();
int32_t InitRecording();
int32_t SetRecordingSampleRate(uint32_t fsHz);
int32_t SetPlayoutSampleRate(uint32_t fsHz);
virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
int32_t RecordingSampleRate() const;
int32_t PlayoutSampleRate() const;
int32_t SetRecordingChannels(uint8_t channels);
int32_t SetPlayoutChannels(uint8_t channels);
virtual int32_t SetRecordingChannels(uint8_t channels);
virtual int32_t SetPlayoutChannels(uint8_t channels);
uint8_t RecordingChannels() const;
uint8_t PlayoutChannels() const;
int32_t SetRecordingChannel(
@ -50,12 +50,13 @@ public:
int32_t RecordingChannel(
AudioDeviceModule::ChannelType& channel) const;
int32_t SetRecordedBuffer(const void* audioBuffer, uint32_t nSamples);
virtual int32_t SetRecordedBuffer(const void* audioBuffer,
uint32_t nSamples);
int32_t SetCurrentMicLevel(uint32_t level);
void SetVQEData(int playDelayMS,
int recDelayMS,
int clockDrift);
int32_t DeliverRecordedData();
virtual void SetVQEData(int playDelayMS,
int recDelayMS,
int clockDrift);
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
virtual int32_t RequestPlayoutData(uint32_t nSamples);