Allowing a custom audio buffer size in WebRtcAudioCapturer

content/renderer/media/webrtc_audio_capturer.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_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "content/child/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/media_stream_constraints_util.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "media/audio/sample_rates.h"
 
 namespace content {
 
 namespace {
 
+// Audio buffer sizes are specified in milliseconds.
+const char kAudioLatency[] = "latencyMs";
+const int kMinAudioLatencyMs = 0;
+const int kMaxAudioLatencyMs = 10000;
+
 // Method to check if any of the data in |audio_source| has energy.
 bool HasDataEnergy(const media::AudioBus& audio_source) {
   for (int ch = 0; ch < audio_source.channels(); ++ch) {
     const float* channel_ptr = audio_source.channel(ch);
     for (int frame = 0; frame < audio_source.frames(); ++frame) {
       if (channel_ptr[frame] != 0)
         return true;
     }
   }
 
@@ skipping 46 matching lines @@
 
     // This can be reentrant so reset |delegate_| before calling out.
     WebRtcLocalAudioTrack* temp = delegate_;
     delegate_ = NULL;
     temp->Stop();
   }
 
   // Wrapper which allows to use std::find_if() when adding and removing
   // sinks to/from the list.
   struct TrackWrapper {
-    TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
+    explicit TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
     bool operator()(
         const scoped_refptr<WebRtcAudioCapturer::TrackOwner>& owner) const {
       return owner->IsEqual(track_);
     }
     WebRtcLocalAudioTrack* track_;
   };
 
  protected:
   virtual ~TrackOwner() {}
 
@@ skipping 89 matching lines @@
            << device_info_.device.input.sample_rate;
   media::AudioSampleRate asr;
   if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected",
                          device_info_.device.input.sample_rate);
   }
 
+  // Initialize the buffer size to zero, which means it wasn't specified.
+  // If it is out of range, we return it to zero.
+  int buffer_size_ms = 0;
+  GetConstraintValueAsInteger(constraints_, kAudioLatency, &buffer_size_ms);
+  if (buffer_size_ms < kMinAudioLatencyMs ||
+      buffer_size_ms > kMaxAudioLatencyMs) {
+    DVLOG(1) << "Ignoring out of range buffer size " << buffer_size_ms;
+    buffer_size_ms = 0;
+  }
+
+  int buffer_size_samples =
+      device_info_.device.input.sample_rate * buffer_size_ms / 1000;

[tommi (sloooow) - chröme, 2015/05/15 06:55:25]

Can we move this calculation into an else clause of the if() statement?  As is,
it's not clear if you actually want to do this calculation when buffer_size_ms
is 0 or if it's a bug (I do see the comment above though, but this calculation
should not be done in that case I think).

[Charlie, 2015/05/15 17:19:32]

Done.

+  DVLOG_IF(1, buffer_size_samples > 0)
+      << "Custom audio buffer size: " << buffer_size_samples << " samples";
+
   // Create and configure the default audio capturing source.
   SetCapturerSourceInternal(
-      AudioDeviceFactory::NewInputDevice(render_frame_id_), channel_layout,
-      static_cast<float>(device_info_.device.input.sample_rate));
+      AudioDeviceFactory::NewInputDevice(render_frame_id_),
+      channel_layout,
+      device_info_.device.input.sample_rate,
+      buffer_size_samples);
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
@@ skipping 64 matching lines @@
     // we have to call StopSource on the MediaStreamSource. This will call
     // MediaStreamAudioSource::DoStopSource which in turn call
     // WebRtcAudioCapturerer::Stop();
     audio_source_->StopSource();
   }
 }
 
 void WebRtcAudioCapturer::SetCapturerSourceInternal(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
-    float sample_rate) {
+    int sample_rate,
+    int buffer_size) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
   {
     base::AutoLock auto_lock(lock_);
     if (source_.get() == source.get())
       return;
 
     source_.swap(old_source);
     source_ = source;
 
     // Reset the flag to allow starting the new source.
     running_ = false;
   }
 
   DVLOG(1) << "Switching to a new capture source.";
   if (old_source.get())
     old_source->Stop();
 
+  // If the buffer size is zero, it has not been specified.
+  // We either default to 10ms, or use the hardware buffer size.
+  if (buffer_size == 0)
+    buffer_size = GetBufferSize(sample_rate);
+
   // Dispatch the new parameters both to the sink(s) and to the new source,
   // also apply the new |constraints|.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
   // bits_per_sample is always 16 for now.
-  int buffer_size = GetBufferSize(sample_rate);
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                channel_layout, sample_rate,
-                                16, buffer_size,
+                                channel_layout,
+                                sample_rate,
+                                16,
+                                buffer_size,
                                 device_info_.device.input.effects);
 
   {
     base::AutoLock auto_lock(lock_);
     // Notify the |audio_processor_| of the new format.
     audio_processor_->OnCaptureFormatChanged(params);
 
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
@@ skipping 28 matching lines @@
   // Do nothing if the current buffer size is the WebRtc native buffer size.
   if (GetBufferSize(input_params.sample_rate()) ==
           input_params.frames_per_buffer()) {
     return;
   }
 
   // Create a new audio stream as source which will open the hardware using
   // WebRtc native buffer size.
   SetCapturerSourceInternal(AudioDeviceFactory::NewInputDevice(render_frame_id),
                             input_params.channel_layout(),
-                            static_cast<float>(input_params.sample_rate()));
+                            input_params.sample_rate(),
+                            0);
 }
 
 void WebRtcAudioCapturer::Start() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
   if (running_ || !source_.get())
     return;
 
   // Start the data source, i.e., start capturing data from the current source.
@@ skipping 202 matching lines @@
     return hardware_buffer_size;
   }
 
   return (sample_rate / 100);
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
-  SetCapturerSourceInternal(source, params.channel_layout(),
-                            static_cast<float>(params.sample_rate()));
+  SetCapturerSourceInternal(source,
+                            params.channel_layout(),
+                            params.sample_rate(),
+                            0);
 }
 
 }  // namespace content