Adding audio network adaptor to AudioEncoderOpus.

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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index d03f2d3cc2440c3d5fa029e91c4dd037fcf54c96..f5c7413dd25ff51beac786b0fb45d66fd67cf7d6 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -12,10 +12,14 @@
 
 #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 {
 
@@ -24,6 +28,7 @@ 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;
@@ -104,13 +109,17 @@ int AudioEncoderOpus::Config::GetBitrateBps() const {
     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_));
@@ -141,15 +150,23 @@ void AudioEncoderOpus::Reset() {
 }
 
 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 {
@@ -192,6 +209,65 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   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,
@@ -226,6 +302,9 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
           });
   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.
@@ -282,7 +361,46 @@ bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
                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