Adding audio network adaptor to AudioEncoderOpus.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 48000;
 const int kMinBitrateBps = 500;
 const int kMaxBitrateBps = 512000;
+constexpr int kSupportedFrameLengths[] = {20, 60};
 
 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderOpus::Config config;
   config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
   config.num_channels = codec_inst.channels;
   config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
   config.payload_type = codec_inst.pltype;
   config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
                                                 : AudioEncoderOpus::kAudio;
   return config;
@@ skipping 60 matching lines @@
 }
 
 int AudioEncoderOpus::Config::GetBitrateBps() const {
   RTC_DCHECK(IsOk());
   if (bitrate_bps)
     return *bitrate_bps;  // Explicitly set value.
   else
     return num_channels == 1 ? 32000 : 64000;  // Default value.
 }
 
-AudioEncoderOpus::AudioEncoderOpus(const Config& config)
-    : packet_loss_rate_(0.0), inst_(nullptr) {
+AudioEncoderOpus::AudioEncoderOpus(
+    const Config& config,
+    std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor)
+    : packet_loss_rate_(0.0),
+      inst_(nullptr),
+      audio_network_adaptor_(std::move(audio_network_adaptor)) {

[kwiberg-webrtc, 2016/10/03 12:48:10]

You still take the ANA as an argument in the constructor *and* destroy/create it
in the code below. You were going to stop doing one of them, right?

[minyue-webrtc, 2016/10/03 12:57:15]

This ctor is only used for integration test. I think I should mention it as a
comment.

   RTC_CHECK(RecreateEncoderInstance(config));
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
-    : AudioEncoderOpus(CreateConfig(codec_inst)) {}
+    : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {}
 
 AudioEncoderOpus::~AudioEncoderOpus() {
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderOpus::NumChannels() const {
@@ skipping 10 matching lines @@
 
 int AudioEncoderOpus::GetTargetBitrate() const {
   return config_.GetBitrateBps();
 }
 
 void AudioEncoderOpus::Reset() {
   RTC_CHECK(RecreateEncoderInstance(config_));
 }
 
 bool AudioEncoderOpus::SetFec(bool enable) {
-  auto conf = config_;
-  conf.fec_enabled = enable;
-  return RecreateEncoderInstance(conf);
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  config_.fec_enabled = enable;
+  return true;
 }
 
 bool AudioEncoderOpus::SetDtx(bool enable) {
-  auto conf = config_;
-  conf.dtx_enabled = enable;
-  return RecreateEncoderInstance(conf);
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  config_.dtx_enabled = enable;
+  return true;
 }
 
 bool AudioEncoderOpus::GetDtx() const {
   return config_.dtx_enabled;
 }
 
 bool AudioEncoderOpus::SetApplication(Application application) {
   auto conf = config_;
   switch (application) {
     case Application::kSpeech:
@@ skipping 22 matching lines @@
   }
 }
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   config_.bitrate_bps = rtc::Optional<int>(
       std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps));
   RTC_DCHECK(config_.IsOk());
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
 }
 
+bool AudioEncoderOpus::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    const Clock* clock) {
+  AudioNetworkAdaptorImpl::Config config;
+  config.clock = clock;
+  audio_network_adaptor_.reset(new AudioNetworkAdaptorImpl(
+      config, ControllerManagerImpl::Create(
+                  config_string, NumChannels(),
+                  rtc::ArrayView<const int>(kSupportedFrameLengths,
+                                            arraysize(kSupportedFrameLengths)),

[kwiberg-webrtc, 2016/10/03 12:48:10]

1. Use the constructor that takes only an array argument. That way, you don't
have to give the size manually and thus can't get it wrong.

2. That constructor is implicit, so you should be able to pass just the array
without mentioning ArrayView at all.

[minyue-webrtc, 2016/10/03 12:57:15]

sure, will do.

+                  num_channels_to_encode_, next_frame_length_ms_,
+                  GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
+  return audio_network_adaptor_.get();
+}
+
+void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
+  audio_network_adaptor_.reset(nullptr);
+}
+
+void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetUplinkBandwidth(uplink_bandwidth_bps);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpus::OnReceivedRtt(int rtt_ms) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetRtt(rtt_ms);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpus::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                                   int max_frame_length_ms) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetReceiverFrameLengthRange(min_frame_length_ms,
+                                                      max_frame_length_ms);
+  ApplyAudioNetworkAdaptor();
+}
+
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
 
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
@@ skipping 14 matching lines @@
                                      config_.num_channels),
                 rtc::saturated_cast<int16_t>(max_encoded_bytes),
                 encoded.data());
 
             RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
 
             return static_cast<size_t>(status);
           });
   input_buffer_.clear();
 
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (info.encoded_bytes > 0);
   info.encoder_type = CodecType::kOpus;
   return info;
 }
 
 size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
   return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
@@ skipping 36 matching lines @@
   RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
   if (config.dtx_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
   } else {
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
+
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
   return true;
 }
 
+void AudioEncoderOpus::SetFrameLength(int frame_length_ms) {
+  next_frame_length_ms_ = frame_length_ms;
+}
+
+void AudioEncoderOpus::SetNumChannelsToEncode(size_t num_channels_to_encode) {
+  RTC_DCHECK_GT(num_channels_to_encode, 0u);
+  RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
+
+  if (num_channels_to_encode_ == num_channels_to_encode)
+    return;
+
+  RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
+  num_channels_to_encode_ = num_channels_to_encode;
+}
+
+void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
+  auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
+  // |audio_network_adaptor_| is supposed to be configured to output all
+  // following parameters.
+  RTC_DCHECK(config.bitrate_bps);
+  RTC_DCHECK(config.frame_length_ms);
+  RTC_DCHECK(config.uplink_packet_loss_fraction);
+  RTC_DCHECK(config.enable_fec);
+  RTC_DCHECK(config.enable_dtx);
+  RTC_DCHECK(config.num_channels);
+
+  RTC_DCHECK(*config.frame_length_ms == 20 || *config.frame_length_ms == 60);
+
+  SetTargetBitrate(*config.bitrate_bps);
+  SetFrameLength(*config.frame_length_ms);
+  SetFec(*config.enable_fec);
+  SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
+  SetDtx(*config.enable_dtx);
+  SetNumChannelsToEncode(*config.num_channels);
+}
+
 }  // namespace webrtc