Index: webrtc/modules/audio_device/include/audio_device_defines.h |
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h |
index b847729f05a6fa543ddaf3bd8056e9af909200d6..ccc263c57b0a6e7ad6ae44a45592cd4cfc4f9994 100644 |
--- a/webrtc/modules/audio_device/include/audio_device_defines.h |
+++ b/webrtc/modules/audio_device/include/audio_device_defines.h |
@@ -66,58 +66,16 @@ class AudioTransport { |
int64_t* elapsed_time_ms, |
int64_t* ntp_time_ms) = 0; |
- // Method to pass captured data directly and unmixed to network channels. |
- // |channel_ids| contains a list of VoE channels which are the |
- // sinks to the capture data. |audio_delay_milliseconds| is the sum of |
- // recording delay and playout delay of the hardware. |current_volume| is |
- // in the range of [0, 255], representing the current microphone analog |
- // volume. |key_pressed| is used by the typing detection. |
- // |need_audio_processing| specify if the data needs to be processed by APM. |
- // Currently WebRtc supports only one APM, and Chrome will make sure only |
- // one stream goes through APM. When |need_audio_processing| is false, the |
- // values of |audio_delay_milliseconds|, |current_volume| and |key_pressed| |
- // will be ignored. |
- // The return value is the new microphone volume, in the range of |0, 255]. |
- // When the volume does not need to be updated, it returns 0. |
- // TODO(xians): Remove this interface after Chrome and Libjingle switches |
- // to OnData(). |
- virtual int OnDataAvailable(const int voe_channels[], |
- size_t number_of_voe_channels, |
- const int16_t* audio_data, |
- int sample_rate, |
- size_t number_of_channels, |
- size_t number_of_frames, |
- int audio_delay_milliseconds, |
- int current_volume, |
- bool key_pressed, |
- bool need_audio_processing) { |
- return 0; |
- } |
- |
- // Method to pass the captured audio data to the specific VoE channel. |
- // |voe_channel| is the id of the VoE channel which is the sink to the |
- // capture data. |
- // TODO(xians): Remove this interface after Libjingle switches to |
- // PushCaptureData(). |
- virtual void OnData(int voe_channel, |
- const void* audio_data, |
- int bits_per_sample, |
- int sample_rate, |
- size_t number_of_channels, |
- size_t number_of_frames) {} |
- |
// Method to push the captured audio data to the specific VoE channel. |
// The data will not undergo audio processing. |
// |voe_channel| is the id of the VoE channel which is the sink to the |
// capture data. |
- // TODO(xians): Make the interface pure virtual after Libjingle |
- // has its implementation. |
virtual void PushCaptureData(int voe_channel, |
const void* audio_data, |
int bits_per_sample, |
int sample_rate, |
size_t number_of_channels, |
- size_t number_of_frames) {} |
+ size_t number_of_frames) = 0; |
// Method to pull mixed render audio data from all active VoE channels. |
// The data will not be passed as reference for audio processing internally. |
@@ -129,7 +87,7 @@ class AudioTransport { |
size_t number_of_frames, |
void* audio_data, |
int64_t* elapsed_time_ms, |
- int64_t* ntp_time_ms) {} |
+ int64_t* ntp_time_ms) = 0; |
protected: |
virtual ~AudioTransport() {} |