Test code for issue 3 channels audio.

content/renderer/media/webrtc_audio_capturer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 313 matching lines @@
   // also apply the new |constraints|.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
   // bits_per_sample is always 16 for now.
   int buffer_size = GetBufferSize(sample_rate);
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 channel_layout, 0, sample_rate,
                                 16, buffer_size,
                                 device_info_.device.input.effects);
 
+  // Hack to test.
+  media::AudioParameters params_msap(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      media::CHANNEL_LAYOUT_STEREO,
+      0, sample_rate, 16, buffer_size,
+      device_info_.device.input.effects);
+
   {
     base::AutoLock auto_lock(lock_);
     // Notify the |audio_processor_| of the new format.
-    audio_processor_->OnCaptureFormatChanged(params);
+    audio_processor_->OnCaptureFormatChanged(params_msap);
 
     MediaAudioConstraints audio_constraints(constraints_,
                                             device_info_.device.input.effects);
     need_audio_processing_ = audio_constraints.NeedsAudioProcessing();
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
 
   if (source.get())
     source->Initialize(params, this, session_id());
@@ skipping 140 matching lines @@
     tracks_.RetrieveAndClearTags(&tracks_to_notify_format);
 
     // Set the flag to turn on the audio processing in PeerConnection level.
     // Note that, we turn off the audio processing in PeerConnection if the
     // processor has already processed the data.
     need_audio_processing = need_audio_processing_ ?
         !MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() : false;
   }
 
   DCHECK(audio_processor_->InputFormat().IsValid());
-  DCHECK_EQ(audio_source->channels(),
-            audio_processor_->InputFormat().channels());
+//  DCHECK_EQ(audio_source->channels(),
+//            audio_processor_->InputFormat().channels());
   DCHECK_EQ(audio_source->frames(),
             audio_processor_->InputFormat().frames_per_buffer());
 
+  scoped_ptr<media::AudioBus> audio_source_2ch =
+      media::AudioBus::CreateWrapper(2);
+  audio_source_2ch->SetChannelData(0, audio_source->channel(0));
+  audio_source_2ch->SetChannelData(1, audio_source->channel(1));
+  audio_source_2ch->set_frames(audio_source->frames());
+
   // Notify the tracks on when the format changes. This will do nothing if
   // |tracks_to_notify_format| is empty.
   media::AudioParameters output_params = audio_processor_->OutputFormat();
+//  LOG(ERROR) << "***** LAYOUT: " << output_params.channel_layout();
   for (TrackList::ItemList::const_iterator it = tracks_to_notify_format.begin();
        it != tracks_to_notify_format.end(); ++it) {
     (*it)->OnSetFormat(output_params);
     (*it)->SetAudioProcessor(audio_processor_);
   }
 
+//  LOG(ERROR) << "***** FRAMES = " << audio_source->frames();
+//  LOG(ERROR) << "***** FRAMES 2ch = " << audio_source_2ch->frames();
+
   if ((base::TimeTicks::Now() - last_audio_level_log_time_).InSeconds() >
           kPowerMonitorLogIntervalSeconds) {
     audio_power_monitor_.Scan(*audio_source, audio_source->frames());
 
     last_audio_level_log_time_ = base::TimeTicks::Now();
 
     std::pair<float, bool> result =
         audio_power_monitor_.ReadCurrentPowerAndClip();
     WebRtcLogMessage(base::StringPrintf(
         "WAC::Capture: current_audio_power=%.2fdBFS.", result.first));
 
     audio_power_monitor_.Reset();
   }
 
   // Push the data to the processor for processing.
-  audio_processor_->PushCaptureData(audio_source);
+//  audio_processor_->PushCaptureData(audio_source);
+  audio_processor_->PushCaptureData(audio_source_2ch.get());
 
   // Process and consume the data in the processor until there is not enough
   // data in the processor.
   int16* output = NULL;
   int new_volume = 0;
   while (audio_processor_->ProcessAndConsumeData(
       audio_delay, current_volume, key_pressed, &new_volume, &output)) {
     // Feed the post-processed data to the tracks.
     for (TrackList::ItemList::const_iterator it = tracks.begin();
          it != tracks.end(); ++it) {
@@ skipping 71 matching lines @@
 
 void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()));
 }
 
 }  // namespace content