Add opus audio codec support in remoting

remoting/codec/audio_encoder_opus.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 "remoting/codec/audio_encoder_opus.h"
+
+#include "base/bind.h"
+#include "base/logging.h"
+#include "base/time.h"
+#include "media/base/audio_bus.h"
+#include "media/base/multi_channel_resampler.h"
+#include "third_party/opus/opus.h"
+
+namespace remoting {
+
+namespace {
+
+// Output 160 kb/s bitrate.
+const int kOutputBitrateBps = 160 * 1024;
+
+// Encoded buffer size.
+const int kFrameDefaultBufferSize = 4096;
+
+// Maximum buffer size we'll allocate when encoding before giving up.
+const int kMaxBufferSize = 65536;
+
+// Opus doesn't support 44100 sampling rate so we always resample to 48kHz.
+const AudioPacket::SamplingRate kOpusSamplingRate =
+    AudioPacket::SAMPLING_RATE_48000;
+
+// Opus supports frame sizes of 2.5, 5, 10, 20, 40 and 60 ms. We use 20 ms
+// frames to balance latency and efficiency.
+const int kFrameSizeMs = 20;
+
+// Number of samples per frame when using default sampling rate.
+const int kFrameSamples =
+    kOpusSamplingRate * kFrameSizeMs / base::Time::kMillisecondsPerSecond;
+
+const AudioPacket::BytesPerSample kBytesPerSample =
+    AudioPacket::BYTES_PER_SAMPLE_2;
+
+bool IsSupportedSampleRate(int rate) {
+  return rate == 44100 || rate == 48000;
+}
+
+}  // namespace
+
+AudioEncoderOpus::AudioEncoderOpus()
+    : sampling_rate_(0),
+      channels_(AudioPacket::CHANNELS_STEREO),
+      encoder_(NULL),
+      frame_size_(0),
+      resampling_data_(NULL),
+      resampling_data_size_(0),
+      resampling_data_pos_(0) {
+}
+
+AudioEncoderOpus::~AudioEncoderOpus() {
+  DestroyEncoder();
+}
+
+void AudioEncoderOpus::InitEncoder() {
+  DCHECK(!encoder_);
+  int error;
+  encoder_ = opus_encoder_create(kOpusSamplingRate, channels_,
+                                 OPUS_APPLICATION_AUDIO, &error);
+  if (!encoder_) {
+    LOG(ERROR) << "Failed to create OPUS encoder. Error code: " << error;
+    return;
+  }
+
+  opus_encoder_ctl(encoder_, OPUS_SET_BITRATE(kOutputBitrateBps));
+
+  frame_size_ = sampling_rate_ * kFrameSizeMs /
+      base::Time::kMillisecondsPerSecond;
+
+  if (sampling_rate_ != kOpusSamplingRate) {
+    resample_buffer_.reset(
+        new char[kFrameSamples * kBytesPerSample * channels_]);
+    resampler_.reset(new media::MultiChannelResampler(

[Wez, 2012/10/22 22:50:21]

This appears in codereview still over-indented; not sure whether it's actually
indented by four bytes in the patch?

[Sergey Ulanov, 2012/10/23 00:43:50]

Looks correctly to me both in rietveld and in my editor (2 spaces relative to
the if 3 lines above).

+        channels_,
+        static_cast<double>(sampling_rate_) / kOpusSamplingRate,
+        base::Bind(&AudioEncoderOpus::FetchBytesToResampler,
+                   base::Unretained(this))));
+    resampler_bus_ = media::AudioBus::Create(channels_, kFrameSamples);
+  }
+
+  // Drop leftover data because it's for different sampling rate.
+  leftover_samples_ = 0;
+  leftover_buffer_size_ =
+      frame_size_ + media::SincResampler::kMaximumLookAheadSize;
+  leftover_buffer_.reset(
+      new int16[leftover_buffer_size_ * channels_]);
+}
+
+void AudioEncoderOpus::DestroyEncoder() {
+  if (encoder_) {
+    opus_encoder_destroy(encoder_);
+    encoder_ = NULL;
+  }
+
+  resampler_.reset();
+}
+
+bool AudioEncoderOpus::ResetForPacket(AudioPacket* packet) {
+  if (packet->channels() != channels_ ||
+      packet->sampling_rate() != sampling_rate_) {
+    DestroyEncoder();
+
+    channels_ = packet->channels();
+    sampling_rate_ = packet->sampling_rate();
+
+    if (channels_ <= 0 || channels_ > 2 ||
+        !IsSupportedSampleRate(sampling_rate_)) {
+      LOG(WARNING) << "Unsupported OPUS parameters: "
+                   << channels_ << " channels with "
+                   << sampling_rate_ << " samples per second.";
+      return false;
+    }
+
+    InitEncoder();
+  }
+
+  return encoder_ != NULL;
+}
+
+void AudioEncoderOpus::FetchBytesToResampler(media::AudioBus* audio_bus) {

[Wez, 2012/10/22 22:50:21]

nit: FetchBytesToResample or PumpBytesToResampler

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+  DCHECK(resampling_data_);
+  int samples_left = (resampling_data_size_ - resampling_data_pos_) /
+      kBytesPerSample / channels_;
+  DCHECK_LE(audio_bus->frames(), samples_left);
+  audio_bus->FromInterleaved(
+      resampling_data_ + resampling_data_pos_,
+      audio_bus->frames(), kBytesPerSample);
+  resampling_data_pos_ += audio_bus->frames() * kBytesPerSample * channels_;
+  DCHECK_LE(resampling_data_pos_, static_cast<int>(resampling_data_size_));
+}
+
+scoped_ptr<AudioPacket> AudioEncoderOpus::Encode(
+    scoped_ptr<AudioPacket> packet) {
+  DCHECK_EQ(AudioPacket::ENCODING_RAW, packet->encoding());
+  DCHECK_EQ(1, packet->data_size());
+  DCHECK_EQ(kBytesPerSample, packet->bytes_per_sample());
+
+  if (!ResetForPacket(packet.get())) {
+    LOG(ERROR) << "Encoder initialization failed";
+    return scoped_ptr<AudioPacket>();
+  }
+
+  int samples_in_packet = packet->data(0).size() / kBytesPerSample / channels_;
+  const int16* next_sample =
+      reinterpret_cast<const int16*>(packet->data(0).data());
+
+  // Create a new packet of encoded data.
+  scoped_ptr<AudioPacket> encoded_packet(new AudioPacket());
+  encoded_packet->set_encoding(AudioPacket::ENCODING_OPUS);
+  encoded_packet->set_sampling_rate(kOpusSamplingRate);
+  encoded_packet->set_channels(channels_);
+
+  int prefetch_samples =
+      resampler_.get() ? media::SincResampler::kMaximumLookAheadSize : 0;
+  int samples_wanted = frame_size_ + prefetch_samples;
+
+  while (leftover_samples_ + samples_in_packet >= samples_wanted) {
+    const int16* pcm_buffer = NULL;
+
+    // Combine the packet with the leftover samples, if any.
+    if (leftover_samples_ > 0) {
+      pcm_buffer = leftover_buffer_.get();
+      int samples_to_copy = samples_wanted - leftover_samples_;
+      memcpy(leftover_buffer_.get() + leftover_samples_ * channels_,
+             next_sample, samples_to_copy * kBytesPerSample * channels_);
+    } else {
+      pcm_buffer = next_sample;
+    }
+
+    // Resample data if necessary.
+    int samples_consumed = 0;
+    if (resampler_.get()) {
+      resampling_data_ = reinterpret_cast<const char*>(pcm_buffer);
+      resampling_data_pos_ = 0;
+      resampling_data_size_ = samples_wanted * channels_ * kBytesPerSample;
+      resampler_->Resample(resampler_bus_.get(), kFrameSamples);
+      resampling_data_ = NULL;
+      samples_consumed = resampling_data_pos_ / channels_ / kBytesPerSample;
+
+      resampler_bus_->ToInterleaved(kFrameSamples, kBytesPerSample,
+                                    resample_buffer_.get());
+      pcm_buffer = reinterpret_cast<int16*>(resample_buffer_.get());
+    } else {
+      samples_consumed = frame_size_;
+    }
+
+    // Initialize output buffer.
+    std::string* data = encoded_packet->add_data();
+    data->resize(kFrameSamples * kBytesPerSample * channels_);
+
+    // Encode.
+    unsigned char* buffer =
+        reinterpret_cast<unsigned char*>(string_as_array(data));
+    int result = opus_encode(encoder_, pcm_buffer, kFrameSamples,
+                             buffer, data->length());
+    if (result < 0) {
+      LOG(ERROR) << "opus_encode() failed with error code: " << result;
+      return scoped_ptr<AudioPacket>();
+    }
+
+    DCHECK_LE(result, static_cast<int>(data->length()));
+    data->resize(result);
+
+    // Cleanup leftover buffer.
+    if (samples_consumed >= leftover_samples_) {
+      samples_consumed -= leftover_samples_;
+      leftover_samples_ = 0;
+      next_sample += samples_consumed * channels_;
+      samples_in_packet -= samples_consumed;
+    } else {
+      leftover_samples_ -= samples_consumed;
+      memmove(leftover_buffer_.get(),
+              leftover_buffer_.get() + samples_consumed * channels_,
+              leftover_samples_ * channels_ * kBytesPerSample);
+    }
+  }
+
+  // Store the leftover samples.
+  if (samples_in_packet > 0) {
+    DCHECK_LE(leftover_samples_ + samples_in_packet, leftover_buffer_size_);
+    memmove(leftover_buffer_.get() + leftover_samples_ * channels_,
+            next_sample, samples_in_packet * kBytesPerSample * channels_);
+    leftover_samples_ += samples_in_packet;
+  }
+
+  // Return NULL if there's nothing in the packet.
+  if (encoded_packet->data_size() == 0)
+    return scoped_ptr<AudioPacket>();
+
+  return encoded_packet.Pass();
+}
+
+}  // namespace remoting