Allow more than 2 input channels in AudioProcessing.

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 94 matching lines @@
   FakeAudioProcessing() : experimental_ns_enabled_(false) {}
 
   WEBRTC_STUB(Initialize, ())
   WEBRTC_STUB(Initialize, (
       int input_sample_rate_hz,
       int output_sample_rate_hz,
       int reverse_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       webrtc::AudioProcessing::ChannelLayout reverse_layout));
+  WEBRTC_STUB(Initialize, (
+      const webrtc::ProcessingConfig& processing_config));
 
   WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
   WEBRTC_STUB(set_sample_rate_hz, (int rate));
   WEBRTC_STUB_CONST(input_sample_rate_hz, ());
   WEBRTC_STUB_CONST(sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
   WEBRTC_STUB_CONST(num_input_channels, ());
   WEBRTC_STUB_CONST(num_output_channels, ());
   WEBRTC_STUB_CONST(num_reverse_channels, ());
   WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
   WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ());
   WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessStream, (
       const float* const* src,
       int samples_per_channel,
       int input_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       int output_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       float* const* dest));
+  WEBRTC_STUB(ProcessStream,
+              (const float* const* src,
+               const webrtc::StreamConfig& input_config,
+               const webrtc::StreamConfig& output_config,
+               float* const* dest));
   WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(AnalyzeReverseStream, (
       const float* const* data,
       int samples_per_channel,
       int sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout layout));
+  WEBRTC_STUB(AnalyzeReverseStream, (
+      const float* const* data,
+      const webrtc::StreamConfig& reverse_config));
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
   WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ());
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
   WEBRTC_STUB(StopDebugRecording, ());
@@ skipping 1115 matching lines @@
   DtmfInfo dtmf_info_;
   webrtc::VoEMediaProcess* media_processor_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_