Adds proper 44.1 support to WebRTC

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright 2013 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_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 83 matching lines @@
     // Map internal volume range of [0.0, 1.0] into [0, 255] used by the
     // webrtc::VoiceEngine.
     microphone_volume_ = static_cast<uint32_t>(volume * kMaxVolumeLevel);
   }
 
   const int channels = number_of_channels;
   DCHECK_LE(channels, input_channels());
   uint32_t new_mic_level = 0;
 
   int samples_per_sec = input_sample_rate();
-  if (samples_per_sec == 44100) {
-    // Even if the hardware runs at 44.1kHz, we use 44.0 internally.
-    samples_per_sec = 44000;
-  }
   const int samples_per_10_msec = (samples_per_sec / 100);
   int bytes_per_sample = input_audio_parameters.bits_per_sample() / 8;
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample;
   int accumulated_audio_samples = 0;
 
   const uint8* audio_byte_buffer = reinterpret_cast<const uint8*>(audio_data);
 
   // Write audio samples in blocks of 10 milliseconds to the registered
   // webrtc::AudioTransport sink. Keep writing until our internal byte
@@ skipping 46 matching lines @@
   {
     base::AutoLock auto_lock(lock_);
     // Store the reported audio delay locally.
     output_delay_ms_ = audio_delay_milliseconds;
   }
 
   const int channels = number_of_channels;
   DCHECK_LE(channels, output_channels());
 
   int samples_per_sec = output_sample_rate();
-  if (samples_per_sec == 44100) {
-    // Even if the hardware runs at 44.1kHz, we use 44.0 internally.
-    samples_per_sec = 44000;
-  }
   int samples_per_10_msec = (samples_per_sec / 100);
   int bytes_per_sample = output_audio_parameters_.bits_per_sample() / 8;
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample;
 
   uint32_t num_audio_samples = 0;
   int accumulated_audio_samples = 0;
 
   // Get audio samples in blocks of 10 milliseconds from the registered
   // webrtc::AudioTransport source. Keep reading until our internal buffer
@@ skipping 311 matching lines @@
     return false;
 
   if (!renderer->Initialize(this))
     return false;
 
   renderer_ = renderer;
   return true;
 }
 
 }  // namespace content