Revert 187936 "Pass more detailed audio hardware configuration i..."

content/renderer/media/webrtc_audio_renderer.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_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 140 matching lines @@
   } else if (sample_rate == 44100) {
     // The resampler in WebRTC does not support 441 as input. We hard code
     // the size to 440 (~0.9977ms) instead and rely on the internal jitter
     // buffer in WebRTC to deal with the resulting drift.
     // TODO(henrika): ensure that WebRTC supports 44100Hz and use 441 instead.
     buffer_size = 440;
   } else {
     return false;
   }
 
-  int channels = ChannelLayoutToChannelCount(channel_layout);
   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                      channel_layout, channels, 0,
-                      sample_rate, 16, buffer_size);
+                      channel_layout, 0, sample_rate, 16, buffer_size);
 
   // Set up audio parameters for the sink, i.e., the native audio output stream.
   // We strive to open up using native parameters to achieve best possible
   // performance and to ensure that no FIFO is needed on the browser side to
   // match the client request. Any mismatch between the source and the sink is
   // taken care of in this class instead using a pull FIFO.
 
   media::AudioParameters sink_params;
 
   buffer_size = hardware_config->GetOutputBufferSize();
   sink_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                    channel_layout, channels, 0, sample_rate, 16, buffer_size);
+                    channel_layout, 0, sample_rate, 16, buffer_size);
 
   // Create a FIFO if re-buffering is required to match the source input with
   // the sink request. The source acts as provider here and the sink as
   // consumer.
   if (source_params.frames_per_buffer() != sink_params.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
              << " to " << sink_params.frames_per_buffer();
     audio_fifo_.reset(new media::AudioPullFifo(
         source_params.channels(),
         source_params.frames_per_buffer(),
@@ skipping 159 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content