Ensures that WebRTC works for device selection using a different sample rate than default

content/renderer/media/webrtc_audio_device_impl.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_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 18 matching lines @@
 }  // namespace
 
 WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
     : ref_count_(0),
       render_loop_(base::MessageLoopProxy::current()),
       audio_transport_callback_(NULL),
       input_delay_ms_(0),
       output_delay_ms_(0),
       last_error_(AudioDeviceModule::kAdmErrNone),
       last_process_time_(base::TimeTicks::Now()),
-      session_id_(0),
       initialized_(false),
       playing_(false),
       agc_is_enabled_(false) {
   DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()";
 }
 
 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() {
   DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()";
   Terminate();
 }
@@ skipping 157 matching lines @@
     // before the new level is effective due to the IPC scheme.
     // During this time, |current_mic_level| will contain "non-valid" values
     // and it might reduce the AGC performance. Measurements on Windows 7 have
     // shown that we might receive old volume levels for one or two callbacks.
     SetMicrophoneVolume(new_mic_level);
   }
 }
 
 void WebRtcAudioDeviceImpl::SetCaptureFormat(
     const media::AudioParameters& params) {
+  DVLOG(1) << "WebRtcAudioDeviceImpl::SetCaptureFormat()";
   input_audio_parameters_ = params;
 }
 
 int32_t WebRtcAudioDeviceImpl::ChangeUniqueId(const int32_t id) {
   NOTIMPLEMENTED();
   return -1;
 }
 
 int32_t WebRtcAudioDeviceImpl::TimeUntilNextProcess() {
   // Calculate the number of milliseconds until this module wants its
@@ skipping 258 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::StartRecording() {
   DCHECK(initialized_);
   DVLOG(1) << "StartRecording()";
   LOG_IF(ERROR, !audio_transport_callback_) << "Audio transport is missing";
   if (!audio_transport_callback_ || !capturer_) {
     return -1;
   }
 
-  if (session_id_ <= 0) {
-    LOG(ERROR) << session_id_ << " is an invalid session id.";
-    return -1;
-  }
-
   if (capturer_->is_recording()) {
     // webrtc::VoiceEngine assumes that it is OK to call Start() twice and
     // that the call is ignored the second time.
     return 0;
   }
 
   start_capture_time_ = base::Time::Now();
 
-  // Specify the session_id which is mapped to a certain device.
-  capturer_->SetDevice(session_id_);
+  // Start capturing using the selected device.
   capturer_->Start();
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::StopRecording() {
   DVLOG(1) << "StopRecording()";
   if (!capturer_ || !capturer_->is_recording()) {
     // webrtc::VoiceEngine assumes that it is OK to call Stop()
     // more than once.
     return 0;
@@ skipping 353 matching lines @@
 int32_t WebRtcAudioDeviceImpl::SetLoudspeakerStatus(bool enable) {
   NOTIMPLEMENTED();
   return -1;
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
-void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
-  session_id_ = session_id;
-}
-
 }  // namespace content