Implementation of 'AudioContext.getOutputTimestamp' method

media/audio/audio_input_device.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 "media/audio/audio_input_device.h"
 
 #include <stdint.h>
 #include <utility>
 
 #include "base/bind.h"
@@ skipping 23 matching lines @@
   AudioThreadCallback(const AudioParameters& audio_parameters,
                       base::SharedMemoryHandle memory,
                       int memory_length,
                       int total_segments,
                       CaptureCallback* capture_callback);
   ~AudioThreadCallback() override;
 
   void MapSharedMemory() override;
 
   // Called whenever we receive notifications about pending data.
-  void Process(uint32_t pending_data) override;
+  void Process(uint32_t pending_data, const AudioTimestamp& timestamp) override;
 
  private:
   int current_segment_id_;
   uint32_t last_buffer_id_;
   ScopedVector<media::AudioBus> audio_buses_;
   CaptureCallback* capture_callback_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioThreadCallback);
 };
 
@@ skipping 240 matching lines @@
   for (int i = 0; i < total_segments_; ++i) {
     media::AudioInputBuffer* buffer =
         reinterpret_cast<media::AudioInputBuffer*>(ptr);
     std::unique_ptr<media::AudioBus> audio_bus =
         media::AudioBus::WrapMemory(audio_parameters_, buffer->audio);
     audio_buses_.push_back(std::move(audio_bus));
     ptr += segment_length_;
   }
 }
 
-void AudioInputDevice::AudioThreadCallback::Process(uint32_t pending_data) {
+void AudioInputDevice::AudioThreadCallback::Process(
+    uint32_t pending_data,
+    const AudioTimestamp& timestamp) {
   // The shared memory represents parameters, size of the data buffer and the
   // actual data buffer containing audio data. Map the memory into this
   // structure and parse out parameters and the data area.
   uint8_t* ptr = static_cast<uint8_t*>(shared_memory_.memory());
   ptr += current_segment_id_ * segment_length_;
   AudioInputBuffer* buffer = reinterpret_cast<AudioInputBuffer*>(ptr);
 
   // Usually this will be equal but in the case of low sample rate (e.g. 8kHz,
   // the buffer may be bigger (on mac at least)).
   DCHECK_GE(buffer->params.size,
@@ skipping 24 matching lines @@
   capture_callback_->Capture(
       audio_bus,
       buffer->params.hardware_delay_bytes / bytes_per_ms_,  // Delay in ms
       buffer->params.volume, buffer->params.key_pressed);
 
   if (++current_segment_id_ >= total_segments_)
     current_segment_id_ = 0;
 }
 
 }  // namespace media