changed MSAP to work with 96000KHZ.

The problem why it did not work is that the FIFO used in MSAP has a size smaller than 10ms buffer before, so it crashed before webrtc::AudioProcessing consumes it.

FYI, previously the APM in Chrome was only enabled for finch experiment on Canary, so the problem does not occur in stable and beta. And I recently enabled it as default.


BUG=378895
TEST= use 96000KHz as sample rate, go to apprtc.appspot.com/?&debug=loopback to make a call.

Review URL: https://codereview.chromium.org/308643004

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@274216 0039d316-1c4b-4281-b951-d872f2087c98

content/renderer/media/media_stream_audio_processor.cc

@@ -64,13 +64,19 @@ class MediaStreamAudioProcessor::MediaStreamAudioConverter
     // |MediaStreamAudioProcessor::capture_converter_|.
     thread_checker_.DetachFromThread();
     audio_converter_.AddInput(this);
+
     // Create and initialize audio fifo and audio bus wrapper.
     // The size of the FIFO should be at least twice of the source buffer size
-    // or twice of the sink buffer size.
+    // or twice of the sink buffer size. Also, FIFO needs to have enough space
+    // to store pre-processed data before passing the data to
+    // webrtc::AudioProcessing, which requires 10ms as packet size.
+    int max_frame_size = std::max(source_params_.frames_per_buffer(),
+                                  sink_params_.frames_per_buffer());
     int buffer_size = std::max(
-        kMaxNumberOfBuffersInFifo * source_params_.frames_per_buffer(),
-        kMaxNumberOfBuffersInFifo * sink_params_.frames_per_buffer());
+        kMaxNumberOfBuffersInFifo * max_frame_size,
+        kMaxNumberOfBuffersInFifo * source_params_.sample_rate() / 100);
     fifo_.reset(new media::AudioFifo(source_params_.channels(), buffer_size));
+
     // TODO(xians): Use CreateWrapper to save one memcpy.
     audio_wrapper_ = media::AudioBus::Create(sink_params_.channels(),
                                              sink_params_.frames_per_buffer());

content/renderer/media/media_stream_audio_processor_unittest.cc

@@ -69,14 +69,15 @@ class MediaStreamAudioProcessorTest : public ::testing::Test {
                                   int expected_output_channels,
                                   int expected_output_buffer_size) {
     // Read the audio data from a file.
+    const media::AudioParameters& params = audio_processor->InputFormat();
     const int packet_size =
-        params_.frames_per_buffer() * 2 * params_.channels();
+        params.frames_per_buffer() * 2 * params.channels();
     const size_t length = packet_size * kNumberOfPacketsForTest;
     scoped_ptr<char[]> capture_data(new char[length]);
     ReadDataFromSpeechFile(capture_data.get(), length);
     const int16* data_ptr = reinterpret_cast<const int16*>(capture_data.get());
     scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create(
-        params_.channels(), params_.frames_per_buffer());
+        params.channels(), params.frames_per_buffer());
     for (int i = 0; i < kNumberOfPacketsForTest; ++i) {
       data_bus->FromInterleaved(data_ptr, data_bus->frames(), 2);
       audio_processor->PushCaptureData(data_bus.get());
@@ -92,7 +93,7 @@ class MediaStreamAudioProcessorTest : public ::testing::Test {
       const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
 #endif
       if (is_aec_enabled) {
-        audio_processor->OnPlayoutData(data_bus.get(), params_.sample_rate(),
+        audio_processor->OnPlayoutData(data_bus.get(), params.sample_rate(),
                                        10);
       }
 
@@ -110,7 +111,7 @@ class MediaStreamAudioProcessorTest : public ::testing::Test {
                   expected_output_buffer_size);
       }
 
-      data_ptr += params_.frames_per_buffer() * params_.channels();
+      data_ptr += params.frames_per_buffer() * params.channels();
     }
   }
 
@@ -350,4 +351,37 @@ TEST_F(MediaStreamAudioProcessorTest, ValidateConstraints) {
   EXPECT_FALSE(audio_constraints.IsValid());
 }
 
+TEST_F(MediaStreamAudioProcessorTest, TestAllSampleRates) {
+  MockMediaConstraintFactory constraint_factory;
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl());
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
+      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+          constraint_factory.CreateWebMediaConstraints(), 0,
+          webrtc_audio_device.get()));
+  EXPECT_TRUE(audio_processor->has_audio_processing());
+
+  static const int kSupportedSampleRates[] =
+      { 8000, 16000, 22050, 32000, 44100, 48000, 88200, 96000 };
+  for (size_t i = 0; i < arraysize(kSupportedSampleRates); ++i) {
+    int buffer_size = (kSupportedSampleRates[i] / 100)  < 128 ?
+        kSupportedSampleRates[i] / 100 : 128;
+    media::AudioParameters params(
+        media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+        media::CHANNEL_LAYOUT_STEREO, kSupportedSampleRates[i], 16,
+        buffer_size);
+    audio_processor->OnCaptureFormatChanged(params);
+    VerifyDefaultComponents(audio_processor);
+
+    ProcessDataAndVerifyFormat(audio_processor,
+                               kAudioProcessingSampleRate,
+                               kAudioProcessingNumberOfChannel,
+                               kAudioProcessingSampleRate / 100);
+  }
+
+  // Set |audio_processor| to NULL to make sure |webrtc_audio_device|
+  // outlives |audio_processor|.
+  audio_processor = NULL;
+}
+
 }  // namespace content