AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo

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 a2497c7862aa748da741d6d127984344f965b60c..edafbbdf39d9f64449395209bde4941a2df8d66f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -29,7 +29,7 @@ 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 = codec_inst.rate;
+  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;
@@ -78,18 +78,32 @@ double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
 
 }  // namespace
 
+AudioEncoderOpus::Config::Config() = default;
+AudioEncoderOpus::Config::Config(const Config&) = default;
+AudioEncoderOpus::Config::~Config() = default;
+auto AudioEncoderOpus::Config::operator=(const Config&) -> Config& = default;

[hlundin-webrtc, 2016/05/18 00:34:47]

I don't understand this syntax. Please, educate me.

[kwiberg-webrtc, 2016/05/18 02:54:28]

It's a trailing return type (see
http://en.cppreference.com/w/cpp/language/function, and search for "trailing").
I'm surprised you haven't heard of it---it's an often used feature that's been
on the Chromium C++11 allowed language features list for almost a week now!

The advantage of using it here is that I get to call the return type Config&
instead of AudioEncoderOpus::Config&, which prevents a line break, making things
much easier to read.

+
 bool AudioEncoderOpus::Config::IsOk() const {
   if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
     return false;
   if (num_channels != 1 && num_channels != 2)
     return false;
-  if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
+  if (bitrate_bps &&
+      (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps))
     return false;
   if (complexity < 0 || complexity > 10)
     return false;
   return true;
 }
 
+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) {
   RTC_CHECK(RecreateEncoderInstance(config));
@@ -119,7 +133,7 @@ size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
 }
 
 int AudioEncoderOpus::GetTargetBitrate() const {
-  return config_.bitrate_bps;
+  return config_.GetBitrateBps();
 }
 
 void AudioEncoderOpus::Reset() {
@@ -168,10 +182,10 @@ void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
 }
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
-  config_.bitrate_bps =
-      std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps);
+  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_.bitrate_bps));
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
@@ -228,7 +242,7 @@ size_t AudioEncoderOpus::SufficientOutputBufferSize() const {
   // Calculate the number of bytes we expect the encoder to produce,
   // then multiply by two to give a wide margin for error.
   const size_t bytes_per_millisecond =
-      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
+      static_cast<size_t>(config_.GetBitrateBps() / (1000 * 8) + 1);
   const size_t approx_encoded_bytes =
       Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
@@ -246,7 +260,7 @@ bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
   input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
                                            config.application));
-  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.bitrate_bps));
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.GetBitrateBps()));
   if (config.fec_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
   } else {