Avoids irregular OnMoreData callbacks on Windows using Core Audio

content/renderer/media/webrtc_audio_device_unittest.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 "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
@@ skipping 27 matching lines @@
     return media::GetAudioHardwareSampleRate();
   }
   virtual int GetAudioInputHardwareSampleRate(
       const std::string& device_id) OVERRIDE {
     return media::GetAudioInputHardwareSampleRate(device_id);
   }
   virtual media::ChannelLayout GetAudioInputHardwareChannelLayout(
       const std::string& device_id) OVERRIDE {
     return media::GetAudioInputHardwareChannelLayout(device_id);
   }
+  virtual int GetAudioOutputBufferSize() OVERRIDE {
+    return media::GetAudioHardwareBufferSize();
+  }
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioUtil);
 };
 
 class AudioUtilNoHardware : public AudioUtilInterface {
  public:
   AudioUtilNoHardware(int output_rate, int input_rate,
                       media::ChannelLayout input_channel_layout)
       : output_rate_(output_rate),
         input_rate_(input_rate),
         input_channel_layout_(input_channel_layout) {
   }
 
   virtual int GetAudioHardwareSampleRate() OVERRIDE {
     return output_rate_;
   }
   virtual int GetAudioInputHardwareSampleRate(
       const std::string& device_id) OVERRIDE {
     return input_rate_;
   }
   virtual media::ChannelLayout GetAudioInputHardwareChannelLayout(
       const std::string& device_id) OVERRIDE {
     return input_channel_layout_;
   }
+  virtual int GetAudioOutputBufferSize() OVERRIDE {
+    return (output_rate_ / 100);
+  }
 
  private:
   int output_rate_;
   int input_rate_;
   media::ChannelLayout input_channel_layout_;
   DISALLOW_COPY_AND_ASSIGN(AudioUtilNoHardware);
 };
 
 // Return true if at least one element in the array matches |value|.
 bool FindElementInArray(int* array, int size, int value) {
   return (std::find(&array[0], &array[0] + size, value) != &array[size]);
 }
 
 // This method returns false if a non-supported rate is detected on the
 // input or output side.
 // TODO(henrika): add support for automatic fallback to Windows Wave audio
 // if a non-supported rate is detected. It is probably better to detect
 // invalid audio settings by actually trying to open the audio streams instead
 // of relying on hard coded conditions.
 bool HardwareSampleRatesAreValid() {
   // These are the currently supported hardware sample rates in both directions.
   // The actual WebRTC client can limit these ranges further depending on
   // platform but this is the maximum range we support today.
   int valid_input_rates[] = {16000, 32000, 44100, 48000, 96000};

[tommi (sloooow) - chröme, 2013/01/31 13:42:08]

nit: make these const

[henrika (OOO until Aug 14), 2013/01/31 14:29:38]

Done.

-  int valid_output_rates[] = {44100, 48000, 96000};
+  int valid_output_rates[] = {16000, 32000, 44100, 48000, 96000};
 
   // Verify the input sample rate.
   int input_sample_rate = GetAudioInputSampleRate();
 
   if (!FindElementInArray(valid_input_rates, arraysize(valid_input_rates),
                           input_sample_rate)) {
     LOG(WARNING) << "Non-supported input sample rate detected.";
     return false;
   }
 
@@ skipping 363 matching lines @@
   EXPECT_EQ(0, file->GetFileDuration(file_path.c_str(), duration,
                                      webrtc::kFileFormatPcm16kHzFile));
   EXPECT_NE(0, duration);
 
   EXPECT_EQ(0, file->StartPlayingFileLocally(ch, file_path.c_str(), false,
                                              webrtc::kFileFormatPcm16kHzFile));
 
   // Play 2 seconds worth of audio and then quit.
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
-                                base::TimeDelta::FromSeconds(2));
+                                base::TimeDelta::FromSeconds(10));
   message_loop_.Run();
 
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 // Uses WebRtcAudioDeviceImpl to play out recorded audio in loopback.
@@ skipping 57 matching lines @@
   scoped_ptr<WebRTCTransportImpl> transport(
       new WebRTCTransportImpl(network.get()));
   EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
   EXPECT_EQ(0, base->StartPlayout(ch));
   EXPECT_EQ(0, base->StartSend(ch));
   renderer->Play();
 
   LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
-                                base::TimeDelta::FromSeconds(2));
+                                base::TimeDelta::FromSeconds(10));
   message_loop_.Run();
 
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 }  // namespace content