| 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..5957adc704a74abe6a44b65583a5ba1d3033fef2 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);
|
|
|
| // 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);
|
|
|
| protected:
|
| virtual ~AudioTransport() {}
|
|
|