Remove old methods in AudioTransport, make it pass a gn build

webrtc/modules/audio_device/include/audio_device_defines.h

 /*
  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 48 matching lines @@
 
   virtual int32_t NeedMorePlayData(const size_t nSamples,
                                    const size_t nBytesPerSample,
                                    const size_t nChannels,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
                                    size_t& nSamplesOut,
                                    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.
   // TODO(xians): Support getting the unmixed render data from specific VoE
   // channel.
   virtual void PullRenderData(int bits_per_sample,
                               int sample_rate,
                               size_t number_of_channels,
                               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() {}
 };
 
 // Helper class for storage of fundamental audio parameters such as sample rate,
 // number of channels, native buffer size etc.
 // Note that one audio frame can contain more than one channel sample and each
 // sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
 // stereo contains 2 * (16/8) = 4 bytes of data.
@@ skipping 58 matching lines @@
  private:
   int sample_rate_;
   size_t channels_;
   size_t frames_per_buffer_;
   size_t frames_per_10ms_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_