Switch AudioRenderSink::Callback to use AudioBus.

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 156 matching lines @@
 
 int32_t WebRtcAudioDeviceImpl::Release() {
   int ret = base::subtle::Barrier_AtomicIncrement(&ref_count_, -1);
   if (ret == 0) {
     delete this;
   }
   return ret;
 }
 
 int WebRtcAudioDeviceImpl::Render(
-    const std::vector<float*>& audio_data,
-    int number_of_frames,
+    media::AudioBus* audio_bus,
     int audio_delay_milliseconds) {
-  DCHECK_LE(number_of_frames, output_buffer_size());
+  DCHECK_LE(audio_bus->frames(), output_buffer_size());
 
   {
     base::AutoLock auto_lock(lock_);
     // Store the reported audio delay locally.
     output_delay_ms_ = audio_delay_milliseconds;
   }
 
-  const int channels = audio_data.size();
+  const int channels = audio_bus->channels();
   DCHECK_LE(channels, output_channels());
 
   int samples_per_sec = output_sample_rate();
   if (samples_per_sec == 44100) {
     // Even if the hardware runs at 44.1kHz, we use 44.0 internally.
     samples_per_sec = 44000;
   }
   int samples_per_10_msec = (samples_per_sec / 100);
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample_;
 
   uint32_t num_audio_samples = 0;
   int accumulated_audio_samples = 0;
 
   char* audio_byte_buffer = reinterpret_cast<char*>(output_buffer_.get());
 
   // Get audio samples in blocks of 10 milliseconds from the registered
   // webrtc::AudioTransport source. Keep reading until our internal buffer
   // is full.
-  while (accumulated_audio_samples < number_of_frames) {
+  while (accumulated_audio_samples < audio_bus->frames()) {
     // Get 10ms and append output to temporary byte buffer.
     audio_transport_callback_->NeedMorePlayData(samples_per_10_msec,
                                                 bytes_per_sample_,
                                                 channels,
                                                 samples_per_sec,
                                                 audio_byte_buffer,
                                                 num_audio_samples);
     accumulated_audio_samples += num_audio_samples;
     audio_byte_buffer += bytes_per_10_msec;
   }
 
   // Deinterleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   for (int channel_index = 0; channel_index < channels; ++channel_index) {
     media::DeinterleaveAudioChannel(
         output_buffer_.get(),
-        audio_data[channel_index],
+        audio_bus->channel(channel_index),
         channels,
         channel_index,
         bytes_per_sample_,
-        number_of_frames);
+        audio_bus->frames());
   }
-  return number_of_frames;
+  return audio_bus->frames();
 }
 
 void WebRtcAudioDeviceImpl::OnRenderError() {
   DCHECK_EQ(MessageLoop::current(), ChildProcess::current()->io_message_loop());
   // TODO(henrika): Implement error handling.
   LOG(ERROR) << "OnRenderError()";
 }
 
-void WebRtcAudioDeviceImpl::Capture(const std::vector<float*>& audio_data,
-                                    int number_of_frames,
+void WebRtcAudioDeviceImpl::Capture(media::AudioBus* audio_bus,
                                     int audio_delay_milliseconds,
                                     double volume) {
-  DCHECK_LE(number_of_frames, input_buffer_size());
+  DCHECK_LE(audio_bus->frames(), input_buffer_size());
 #if defined(OS_WIN) || defined(OS_MACOSX)
   DCHECK_LE(volume, 1.0);
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
   // We have a special situation on Linux where the microphone volume can be
   // "higher than maximum". The input volume slider in the sound preference
   // allows the user to set a scaling that is higher than 100%. It means that
   // even if the reported maximum levels is N, the actual microphone level can
   // go up to 1.5*N and that corresponds to a normalized |volume| of 1.5.
   DCHECK_LE(volume, 1.5);
 #endif
 
   int output_delay_ms = 0;
   {
     base::AutoLock auto_lock(lock_);
     // Store the reported audio delay locally.
     input_delay_ms_ = audio_delay_milliseconds;
     output_delay_ms = output_delay_ms_;
   }
 
-  const int channels = audio_data.size();
+  const int channels = audio_bus->channels();
   DCHECK_LE(channels, input_channels());
   uint32_t new_mic_level = 0;
 
   // Interleave, scale, and clip input to int and store result in
   // a local byte buffer.
-  media::InterleaveFloatToInt(audio_data,
+  media::InterleaveFloatToInt(audio_bus,
                               input_buffer_.get(),
-                              number_of_frames,
+                              audio_bus->frames(),
                               input_audio_parameters_.bits_per_sample() / 8);
 
   int samples_per_sec = input_sample_rate();
   if (samples_per_sec == 44100) {
     // Even if the hardware runs at 44.1kHz, we use 44.0 internally.
     samples_per_sec = 44000;
   }
   const int samples_per_10_msec = (samples_per_sec / 100);
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample_;
   int accumulated_audio_samples = 0;
 
   char* audio_byte_buffer = reinterpret_cast<char*>(input_buffer_.get());
 
   // Map internal volume range of [0.0, 1.0] into [0, 255] used by the
   // webrtc::VoiceEngine.
   uint32_t current_mic_level = static_cast<uint32_t>(volume * kMaxVolumeLevel);
 
   // Write audio samples in blocks of 10 milliseconds to the registered
   // webrtc::AudioTransport sink. Keep writing until our internal byte
   // buffer is empty.
-  while (accumulated_audio_samples < number_of_frames) {
+  while (accumulated_audio_samples < audio_bus->frames()) {
     // Deliver 10ms of recorded 16-bit linear PCM audio.
     audio_transport_callback_->RecordedDataIsAvailable(
         audio_byte_buffer,
         samples_per_10_msec,
         bytes_per_sample_,
         channels,
         samples_per_sec,
         input_delay_ms_ + output_delay_ms,
         0,  // TODO(henrika): |clock_drift| parameter is not utilized today.
         current_mic_level,
@@ skipping 863 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }