Reland: Replace AudioFifo with a deque of AudioBus in ATR::AudioEncoder

content/renderer/media_recorder/audio_track_recorder.cc

 // Copyright 2015 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_recorder/audio_track_recorder.h"
 
 #include <stdint.h>
 #include <utility>
 
 #include "base/bind.h"
 #include "base/macros.h"
 #include "base/stl_util.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_converter.h"
-#include "media/base/audio_fifo.h"
 #include "media/base/audio_parameters.h"
 #include "media/base/audio_sample_types.h"
 #include "media/base/bind_to_current_loop.h"
 #include "third_party/opus/src/include/opus.h"
 
 // Note that this code follows the Chrome media convention of defining a "frame"
 // as "one multi-channel sample" as opposed to another common definition meaning
 // "a chunk of samples". Here this second definition of "frame" is called a
 // "buffer"; so what might be called "frame duration" is instead "buffer
 // duration", and so on.
@@ skipping 93 matching lines @@
   base::ThreadChecker encoder_thread_checker_;
 
   // Track Audio (ingress) and Opus encoder input parameters, respectively. They
   // only differ in their sample_rate() and frames_per_buffer(): output is
   // 48ksamples/s and 2880, respectively.
   media::AudioParameters input_params_;
   media::AudioParameters output_params_;
 
   // Sampling rate adapter between an OpusEncoder supported and the provided.
   std::unique_ptr<media::AudioConverter> converter_;
-  std::unique_ptr<media::AudioFifo> fifo_;
+  std::deque<std::unique_ptr<media::AudioBus>> audio_bus_queue_;
 
   // Buffer for passing AudioBus data to OpusEncoder.
   std::unique_ptr<float[]> buffer_;
 
   // While |paused_|, AudioBuses are not encoded.
   bool paused_;
 
+  int frames_queued_;
+
   OpusEncoder* opus_encoder_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioEncoder);
 };
 
 AudioTrackRecorder::AudioEncoder::AudioEncoder(
     const OnEncodedAudioCB& on_encoded_audio_cb,
     int32_t bits_per_second)
     : on_encoded_audio_cb_(on_encoded_audio_cb),
       bits_per_second_(bits_per_second),
       paused_(false),
+      frames_queued_(0),
       opus_encoder_(nullptr) {
   // AudioEncoder is constructed on the thread that ATR lives on, but should
   // operate only on the encoder thread after that. Reset
   // |encoder_thread_checker_| here, as the next call to CalledOnValidThread()
   // will be from the encoder thread.
   encoder_thread_checker_.DetachFromThread();
 }
 
 AudioTrackRecorder::AudioEncoder::~AudioEncoder() {
   // We don't DCHECK that we're on the encoder thread here, as it should have
   // already been deleted at this point.
   DestroyExistingOpusEncoder();
 }
 
 void AudioTrackRecorder::AudioEncoder::OnSetFormat(
     const media::AudioParameters& input_params) {
   DVLOG(1) << __func__;
   DCHECK(encoder_thread_checker_.CalledOnValidThread());
   if (input_params_.Equals(input_params))
     return;
 
   DestroyExistingOpusEncoder();
 
   if (!input_params.IsValid()) {
     DLOG(ERROR) << "Invalid params: " << input_params.AsHumanReadableString();
     return;
   }
+
   input_params_ = input_params;
   input_params_.set_frames_per_buffer(input_params_.sample_rate() *
                                       kOpusPreferredBufferDurationMs /
                                       base::Time::kMillisecondsPerSecond);
 
   // third_party/libopus supports up to 2 channels (see implementation of
   // opus_encoder_create()): force |output_params_| to at most those.
   output_params_ = media::AudioParameters(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
       media::GuessChannelLayout(std::min(input_params_.channels(), 2)),
       kOpusPreferredSamplingRate,
       input_params_.bits_per_sample(),
       kOpusPreferredFramesPerBuffer);
   DVLOG(1) << "|input_params_|:" << input_params_.AsHumanReadableString()
            << " -->|output_params_|:" << output_params_.AsHumanReadableString();
 
   converter_.reset(new media::AudioConverter(input_params_, output_params_,
                                              false /* disable_fifo */));
   converter_->AddInput(this);
   converter_->PrimeWithSilence();
 
-  fifo_.reset(new media::AudioFifo(
-      input_params_.channels(),
-      kMaxNumberOfFifoBuffers * input_params_.frames_per_buffer()));
+  frames_queued_ = 0;
+  audio_bus_queue_.clear();
 
   buffer_.reset(new float[output_params_.channels() *
                           output_params_.frames_per_buffer()]);
 
   // Initialize OpusEncoder.
   int opus_result;
   opus_encoder_ = opus_encoder_create(output_params_.sample_rate(),
                                       output_params_.channels(),
                                       OPUS_APPLICATION_AUDIO,
                                       &opus_result);
@@ skipping 20 matching lines @@
     std::unique_ptr<media::AudioBus> input_bus,
     const base::TimeTicks& capture_time) {
   DVLOG(3) << __func__ << ", #frames " << input_bus->frames();
   DCHECK(encoder_thread_checker_.CalledOnValidThread());
   DCHECK_EQ(input_bus->channels(), input_params_.channels());
   DCHECK(!capture_time.is_null());
   DCHECK(converter_);
 
   if (!is_initialized() || paused_)
     return;
-  // TODO(mcasas): Consider using a std::deque<std::unique_ptr<AudioBus>>
-  // instead of
-  // an AudioFifo, to avoid copying data needlessly since we know the sizes of
-  // both input and output and they are multiples.
-  fifo_->Push(input_bus.get());
+
+  frames_queued_ += input_bus->frames();
+  audio_bus_queue_.push_back(std::move(input_bus));
+
+  const int max_frame_limit =
+      kMaxNumberOfFifoBuffers * input_params_.frames_per_buffer();
+  while (frames_queued_ > max_frame_limit) {
+    frames_queued_ -= audio_bus_queue_.front()->frames();
+    audio_bus_queue_.pop_front();
+  }
 
   // Wait to have enough |input_bus|s to guarantee a satisfactory conversion.
-  while (fifo_->frames() >= input_params_.frames_per_buffer()) {
+  while (frames_queued_ >= input_params_.frames_per_buffer()) {
     std::unique_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(
         output_params_.channels(), kOpusPreferredFramesPerBuffer);
     converter_->Convert(audio_bus.get());
     audio_bus->ToInterleaved<media::Float32SampleTypeTraits>(
         audio_bus->frames(), buffer_.get());
 
     std::unique_ptr<std::string> encoded_data(new std::string());
     if (DoEncode(opus_encoder_, buffer_.get(), kOpusPreferredFramesPerBuffer,
                  encoded_data.get())) {
       const base::TimeTicks capture_time_of_first_sample =
           capture_time -
-          base::TimeDelta::FromMicroseconds(fifo_->frames() *
+          base::TimeDelta::FromMicroseconds(frames_queued_ *
                                             base::Time::kMicrosecondsPerSecond /
                                             input_params_.sample_rate());
       on_encoded_audio_cb_.Run(output_params_, std::move(encoded_data),
                                capture_time_of_first_sample);
     }
   }
 }
 
 double AudioTrackRecorder::AudioEncoder::ProvideInput(
     media::AudioBus* audio_bus,
     uint32_t frames_delayed) {
-  fifo_->Consume(audio_bus, 0, audio_bus->frames());
+  audio_bus->Zero();
+  if (audio_bus_queue_.empty())
+    return 0.0;
+
+  frames_queued_ -= audio_bus->frames();
+  audio_bus_queue_.front()->CopyTo(audio_bus);
+  audio_bus_queue_.pop_front();
   return 1.0;  // Return volume greater than zero to indicate we have more data.
 }
 
 void AudioTrackRecorder::AudioEncoder::DestroyExistingOpusEncoder() {
   // We don't DCHECK that we're on the encoder thread here, as this could be
   // called from the dtor (main thread) or from OnSetFormat() (encoder thread).
   if (opus_encoder_) {
     opus_encoder_destroy(opus_encoder_);
     opus_encoder_ = nullptr;
   }
@@ skipping 55 matching lines @@
 }
 
 void AudioTrackRecorder::Resume() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   DCHECK(encoder_);
   encoder_thread_.task_runner()->PostTask(
       FROM_HERE, base::Bind(&AudioEncoder::set_paused, encoder_, false));
 }
 
 }  // namespace content