Adds support for 16kHz input sample rate and mono channel config. in WebRTC.

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright (c) 2011 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/string_util.h"
 #include "base/win/windows_version.h"
 #include "content/renderer/media/audio_hardware.h"
@@ skipping 282 matching lines @@
   int output_sample_rate =
       static_cast<int>(audio_hardware::GetOutputSampleRate());
   DVLOG(1) << "Audio output hardware sample rate: " << output_sample_rate;
 
   // Ask the browser for the default audio input hardware sample-rate.
   // This request is based on a synchronous IPC message.
   int input_sample_rate =
       static_cast<int>(audio_hardware::GetInputSampleRate());
   DVLOG(1) << "Audio input hardware sample rate: " << input_sample_rate;
 
-  int input_channels = 0;
+  // Ask the browser for the default number of audio input channels.
+  // This request is based on a synchronous IPC message.
+  int input_channels = audio_hardware::GetInputChannelCount();
+  DVLOG(1) << "Audio input hardware channels: " << input_channels;
+
   int output_channels = 0;
 
   size_t input_buffer_size = 0;
   size_t output_buffer_size = 0;
 
 // Windows
 #if defined(OS_WIN)
-  if (input_sample_rate != 48000 && input_sample_rate != 44100) {
-    DLOG(ERROR) << "Only 48 and 44.1kHz input rates are supported on Windows.";
+  if (input_sample_rate != 48000 && input_sample_rate != 44100 &&
+      input_sample_rate != 32000 && input_sample_rate != 16000) {
+    DLOG(ERROR) << "Only 48, 44.1, 32 and 16kHz input rates are supported.";
     return -1;
   }
   if (output_sample_rate != 48000 && output_sample_rate != 44100) {
-    DLOG(ERROR) << "Only 48 and 44.1kHz output rates are supported on Windows.";
+    DLOG(ERROR) << "Only 48 and 44.1kHz output rates are supported.";
     return -1;
   }
 
-  // Use stereo recording on Windows since low-latency Core Audio (WASAPI)
-  // does not support mono.
-  input_channels = 2;
-
-  // Use stereo rendering on Windows to make input and output sides
-  // symmetric. WASAPI supports both stereo and mono.
+  // Always use stereo rendering on Windows.
   output_channels = 2;
 
   // Capture side: AUDIO_PCM_LOW_LATENCY is based on the Core Audio (WASAPI)
   // API which was introduced in Windows Vista. For lower Windows versions,
   // a callback-driven Wave implementation is used instead. An input buffer
   // size of 10ms works well for both these implementations.
 
   // Use different buffer sizes depending on the current hardware sample rate.
-  if (input_sample_rate == 48000) {
-    input_buffer_size = 480;
-  } else {
+  if (input_sample_rate == 44100) {
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
     // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
     input_buffer_size = 440;
+  } else {
+    input_buffer_size = (input_sample_rate / 100);
   }
 
   // Render side: AUDIO_PCM_LOW_LATENCY is based on the Core Audio (WASAPI)
   // API which was introduced in Windows Vista. For lower Windows versions,
   // a callback-driven Wave implementation is used instead. An output buffer
   // size of 10ms works well for WASAPI but 30ms is needed for Wave.
 
   // Use different buffer sizes depending on the current hardware sample rate.
   if (output_sample_rate == 48000) {
     output_buffer_size = 480;
   } else {
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
     // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
     // TODO(henrika): figure out why we seem to need 20ms here for glitch-
     // free audio.
     output_buffer_size = 2 * 440;
   }
 
   // Windows XP and lower can't cope with 10 ms output buffer size.
   // It must be extended to 30 ms (60 ms will be used internally by WaveOut).
-  if (base::win::GetVersion() <= base::win::VERSION_XP) {
+  if (!media::IsWASAPISupported()) {
     output_buffer_size = 3 * output_buffer_size;
     DLOG(WARNING) << "Extending the output buffer size by a factor of three "
                   << "since Windows XP has been detected.";
   }
 
 // Mac OS X
 #elif defined(OS_MACOSX)
-  if (input_sample_rate != 48000 && input_sample_rate != 44100) {
-    DLOG(ERROR) << "Only 48 and 44.1kHz input rates are supported on Mac OSX.";
+  if (input_sample_rate != 48000 && input_sample_rate != 44100 &&
+      input_sample_rate != 32000 && input_sample_rate != 16000) {

[scherkus (not reviewing), 2012/01/18 18:19:38]

your initial CL comment says only windows is supported -- does this need to be
reverted?

[henrika (OOO until Aug 14), 2012/01/19 09:42:24]

You might be referring to "First patch works for Windows only. Mac OS X is
next." What I meant to describe was that I started out adding support for
Windows and then added Mac OSX, in the same CL. Might not have been clear. 

Can I rewrite old comments?

+    DLOG(ERROR) << "Only 48, 44.1, 32 and 16kHz input rates are supported.";
     return -1;
   }
   if (output_sample_rate != 48000 && output_sample_rate != 44100) {
     DLOG(ERROR) << "Only 48 and 44.1kHz output rates are supported on Mac OSX.";
     return -1;
   }
 
-  input_channels = 1;
   output_channels = 1;
 
   // Capture side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback-
   // driven Core Audio implementation. Tests have shown that 10ms is a suitable
   // frame size to use, both for 48kHz and 44.1kHz.
 
   // Use different buffer sizes depending on the current hardware sample rate.
-  if (input_sample_rate == 48000) {
-    input_buffer_size = 480;
-  } else {
+  if (input_sample_rate == 44100) {
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
     // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
     input_buffer_size = 440;
+  } else {
+    input_buffer_size = (input_sample_rate / 100);
   }
 
   // Render side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback-
   // driven Core Audio implementation. Tests have shown that 10ms is a suitable
   // frame size to use, both for 48kHz and 44.1kHz.
 
   // Use different buffer sizes depending on the current hardware sample rate.
   if (output_sample_rate == 48000) {
     output_buffer_size = 480;
   } else {
@@ skipping 210 matching lines @@
   playing_ = false;
   return 0;
 }
 
 bool WebRtcAudioDeviceImpl::Playing() const {
   return playing_;
 }
 
 int32_t WebRtcAudioDeviceImpl::StartRecording() {
   DVLOG(1) << "StartRecording()";
-#if defined(OS_MACOSX)
-  DLOG(WARNING) << "Real-time recording is not yet fully supported on Mac OS X";
-#endif
   LOG_IF(ERROR, !audio_transport_callback_) << "Audio transport is missing";
   if (!audio_transport_callback_) {
     LOG(ERROR) << "Audio transport is missing";
     return -1;
   }
 
   if (session_id_ <= 0) {
     LOG(WARNING) << session_id_ << " is an invalid session id.";
     // TODO(xians): enable the return -1 when MediaStreamManager can handle
     // AudioInputDeviceManager.
@@ skipping 349 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }