Opus implementation of the AudioEncoderFactoryTemplate API

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 be32aef152e80e05feed810d415f5718ca04a869..9ebb63d8497ceabd2907720b1ab95482a9b2c2e0 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -19,6 +19,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/numerics/exp_filter.h"
 #include "webrtc/base/protobuf_utils.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/safe_minmax.h"
 #include "webrtc/base/string_to_number.h"
@@ -48,11 +49,6 @@ constexpr int kOpusBitrateNbBps = 12000;
 constexpr int kOpusBitrateWbBps = 20000;
 constexpr int kOpusBitrateFbBps = 32000;
 
-// Opus API allows a min bitrate of 500bps, but Opus documentation suggests
-// bitrate should be in the range of 6000 to 510000, inclusive.
-constexpr int kOpusMinBitrateBps = 6000;
-constexpr int kOpusMaxBitrateBps = 510000;
-
 constexpr int kSampleRateHz = 48000;
 constexpr int kDefaultMaxPlaybackRate = 48000;
 
@@ -133,8 +129,8 @@ int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
       return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
     }
   }();
-  RTC_DCHECK_GE(bitrate, kOpusMinBitrateBps);
-  RTC_DCHECK_LE(bitrate, kOpusMaxBitrateBps);
+  RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
+  RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
   return bitrate;
 }
 
@@ -150,7 +146,8 @@ int CalculateBitrate(int max_playback_rate_hz,
     const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
     if (bitrate) {
       const int chosen_bitrate =
-          std::max(kOpusMinBitrateBps, std::min(*bitrate, kOpusMaxBitrateBps));
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
       if (bitrate != chosen_bitrate) {
         LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
                         << " clamped to " << chosen_bitrate;
@@ -195,7 +192,7 @@ int GetFrameSizeMs(const SdpAudioFormat& format) {
     return *(std::end(kOpusSupportedFrameLengths) - 1);
   }
 
-  return AudioEncoderOpus::Config::kDefaultFrameSizeMs;
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
 }
 
 void FindSupportedFrameLengths(int min_frame_length_ms,
@@ -211,8 +208,39 @@ void FindSupportedFrameLengths(int min_frame_length_ms,
   RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
 }
 
+int GetBitrateBps(const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return *config.bitrate_bps;
+}
+
 }  // namespace
 
+void AudioEncoderOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {
+      "opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(48000, config.num_channels, *config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  return info;
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type) {
+  RTC_DCHECK(config.IsOk());
+  return rtc::MakeUnique<AudioEncoderOpus>(config, payload_type);
+}
+
 rtc::Optional<AudioCodecInfo> AudioEncoderOpus::QueryAudioEncoder(
     const SdpAudioFormat& format) {
   if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
@@ -221,8 +249,9 @@ rtc::Optional<AudioCodecInfo> AudioEncoderOpus::QueryAudioEncoder(
     const int bitrate =
         CalculateBitrate(GetMaxPlaybackRate(format), num_channels,
                          GetFormatParameter(format, "maxaveragebitrate"));
-    AudioCodecInfo info(48000, num_channels, bitrate, kOpusMinBitrateBps,
-                        kOpusMaxBitrateBps);
+    AudioCodecInfo info(48000, num_channels, bitrate,
+                        AudioEncoderOpusConfig::kMinBitrateBps,
+                        AudioEncoderOpusConfig::kMaxBitrateBps);
     info.allow_comfort_noise = false;
     info.supports_network_adaption = true;
 
@@ -231,27 +260,36 @@ rtc::Optional<AudioCodecInfo> AudioEncoderOpus::QueryAudioEncoder(
   return rtc::Optional<AudioCodecInfo>();
 }
 
-AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
+AudioEncoderOpusConfig AudioEncoderOpus::CreateConfig(
+    int payload_type,
+    const SdpAudioFormat& format) {
+  auto opt_config = SdpToConfig(format);
+  RTC_CHECK(opt_config);
+  opt_config->payload_type = payload_type;
+  return *opt_config;
+}
+
+AudioEncoderOpusConfig AudioEncoderOpus::CreateConfig(
     const CodecInst& codec_inst) {
-  AudioEncoderOpus::Config config;
+  AudioEncoderOpusConfig 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;
+  config.application = config.num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
   config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
-  config.low_rate_complexity = 9;
-#endif
   return config;
 }
 
-AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
-    int payload_type,
+rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
     const SdpAudioFormat& format) {
-  AudioEncoderOpus::Config config;
+  if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
+      format.clockrate_hz != 48000 || format.num_channels != 2) {
+    return rtc::Optional<AudioEncoderOpusConfig>();
+  }
 
+  AudioEncoderOpusConfig config;
   config.num_channels = GetChannelCount(format);
   config.frame_size_ms = GetFrameSizeMs(format);
   config.max_playback_rate_hz = GetMaxPlaybackRate(format);
@@ -261,16 +299,14 @@ AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
   config.bitrate_bps = rtc::Optional<int>(
       CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
                        GetFormatParameter(format, "maxaveragebitrate")));
-  config.payload_type = payload_type;
-  config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
-                                                : AudioEncoderOpus::kAudio;
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
-  config.low_rate_complexity = 9;
-#endif
+  config.application = config.num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
 
   constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
   constexpr int kMaxANAFrameLength =
       kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
+
   // For now, minptime and maxptime are only used with ANA. If ptime is outside
   // of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
   // if ANA was to be used when setting up the config, and adjust accordingly.
@@ -281,7 +317,25 @@ AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
 
   FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
                             &config.supported_frame_lengths_ms);
-  return config;
+  RTC_DCHECK(config.IsOk());
+  return rtc::Optional<AudioEncoderOpusConfig>(config);
+}
+
+rtc::Optional<int> AudioEncoderOpus::GetNewComplexity(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  const int bitrate_bps = GetBitrateBps(config);
+  if (bitrate_bps >= config.complexity_threshold_bps -
+                         config.complexity_threshold_window_bps &&
+      bitrate_bps <= config.complexity_threshold_bps +
+                         config.complexity_threshold_window_bps) {
+    // Within the hysteresis window; make no change.
+    return rtc::Optional<int>();
+  } else {
+    return rtc::Optional<int>(bitrate_bps <= config.complexity_threshold_bps
+                                  ? config.low_rate_complexity
+                                  : config.complexity);
+  }
 }
 
 class AudioEncoderOpus::PacketLossFractionSmoother {
@@ -311,58 +365,16 @@ class AudioEncoderOpus::PacketLossFractionSmoother {
   rtc::ExpFilter smoother_;
 };
 
-AudioEncoderOpus::Config::Config() {
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
-  low_rate_complexity = 9;
-#endif
-}
-AudioEncoderOpus::Config::Config(const Config&) = default;
-AudioEncoderOpus::Config::~Config() = default;
-auto AudioEncoderOpus::Config::operator=(const Config&) -> Config& = default;
-
-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 &&
-      (*bitrate_bps < kOpusMinBitrateBps || *bitrate_bps > kOpusMaxBitrateBps))
-    return false;
-  if (complexity < 0 || complexity > 10)
-    return false;
-  if (low_rate_complexity < 0 || low_rate_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.
-}
-
-rtc::Optional<int> AudioEncoderOpus::Config::GetNewComplexity() const {
-  RTC_DCHECK(IsOk());
-  const int bitrate_bps = GetBitrateBps();
-  if (bitrate_bps >=
-          complexity_threshold_bps - complexity_threshold_window_bps &&
-      bitrate_bps <=
-          complexity_threshold_bps + complexity_threshold_window_bps) {
-    // Within the hysteresis window; make no change.
-    return rtc::Optional<int>();
-  }
-  return bitrate_bps <= complexity_threshold_bps
-             ? rtc::Optional<int>(low_rate_complexity)
-             : rtc::Optional<int>(complexity);
-}
+AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config)
+    : AudioEncoderOpus(config, config.payload_type) {}
 
 AudioEncoderOpus::AudioEncoderOpus(
-    const Config& config,
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
     AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
     std::unique_ptr<SmoothingFilter> bitrate_smoother)
-    : send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
+    : payload_type_(payload_type),
+      send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
           "WebRTC-SendSideBwe-WithOverhead")),
       packet_loss_rate_(0.0),
       inst_(nullptr),
@@ -379,15 +391,21 @@ AudioEncoderOpus::AudioEncoderOpus(
           ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
               // We choose 5sec as initial time constant due to empirical data.
               new SmoothingFilterImpl(5000))) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  // Sanity check of the redundant payload type field that we want to get rid
+  // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
+
   RTC_CHECK(RecreateEncoderInstance(config));
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
-    : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {}
+    : AudioEncoderOpus(CreateConfig(codec_inst), codec_inst.pltype) {}
 
 AudioEncoderOpus::AudioEncoderOpus(int payload_type,
                                    const SdpAudioFormat& format)
-    : AudioEncoderOpus(CreateConfig(payload_type, format), nullptr) {}
+    : AudioEncoderOpus(*SdpToConfig(format), payload_type) {}
 
 AudioEncoderOpus::~AudioEncoderOpus() {
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
@@ -410,7 +428,7 @@ size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
 }
 
 int AudioEncoderOpus::GetTargetBitrate() const {
-  return config_.GetBitrateBps();
+  return GetBitrateBps(config_);
 }
 
 void AudioEncoderOpus::Reset() {
@@ -445,10 +463,10 @@ bool AudioEncoderOpus::SetApplication(Application application) {
   auto conf = config_;
   switch (application) {
     case Application::kSpeech:
-      conf.application = AudioEncoderOpus::kVoip;
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
       break;
     case Application::kAudio:
-      conf.application = AudioEncoderOpus::kAudio;
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
       break;
   }
   return RecreateEncoderInstance(conf);
@@ -523,9 +541,10 @@ void AudioEncoderOpus::OnReceivedUplinkBandwidth(
     }
     const int overhead_bps = static_cast<int>(
         *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
-    SetTargetBitrate(std::min(
-        kOpusMaxBitrateBps,
-        std::max(kOpusMinBitrateBps, target_audio_bitrate_bps - overhead_bps)));
+    SetTargetBitrate(
+        std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
+                 std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                          target_audio_bitrate_bps - overhead_bps)));
   } else {
     SetTargetBitrate(target_audio_bitrate_bps);
   }
@@ -597,7 +616,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
   config_.frame_size_ms = next_frame_length_ms_;
 
   info.encoded_timestamp = first_timestamp_in_buffer_;
-  info.payload_type = config_.payload_type;
+  info.payload_type = 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;
@@ -616,7 +635,7 @@ 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_.GetBitrateBps() / (1000 * 8) + 1);
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
   const size_t approx_encoded_bytes =
       Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
@@ -625,7 +644,8 @@ size_t AudioEncoderOpus::SufficientOutputBufferSize() const {
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.
-bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
+bool AudioEncoderOpus::RecreateEncoderInstance(
+    const AudioEncoderOpusConfig& config) {
   if (!config.IsOk())
     return false;
   config_ = config;
@@ -633,9 +653,13 @@ bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
     RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
   input_buffer_.clear();
   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.GetBitrateBps()));
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
+                      &inst_, config.num_channels,
+                      config.application ==
+                              AudioEncoderOpusConfig::ApplicationMode::kVoip
+                          ? 0
+                          : 1));
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config)));
   if (config.fec_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
   } else {
@@ -645,7 +669,7 @@ bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
       0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
   // Use the default complexity if the start bitrate is within the hysteresis
   // window.
-  complexity_ = config.GetNewComplexity().value_or(config.complexity);
+  complexity_ = GetNewComplexity(config).value_or(config.complexity);
   RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   if (config.dtx_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
@@ -692,10 +716,11 @@ void AudioEncoderOpus::SetProjectedPacketLossRate(float fraction) {
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
-      bits_per_second, kOpusMinBitrateBps, kOpusMaxBitrateBps));
+      bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+      AudioEncoderOpusConfig::kMaxBitrateBps));
   RTC_DCHECK(config_.IsOk());
-  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
-  const auto new_complexity = config_.GetNewComplexity();
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
+  const auto new_complexity = GetNewComplexity(config_);
   if (new_complexity && complexity_ != *new_complexity) {
     complexity_ = *new_complexity;
     RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
@@ -728,11 +753,11 @@ AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
   AudioNetworkAdaptorImpl::Config config;
   config.event_log = event_log;
   return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
-      config,
-      ControllerManagerImpl::Create(
-          config_string, NumChannels(), supported_frame_lengths_ms(),
-          kOpusMinBitrateBps, num_channels_to_encode_, next_frame_length_ms_,
-          GetTargetBitrate(), config_.fec_enabled, GetDtx())));
+      config, ControllerManagerImpl::Create(
+                  config_string, NumChannels(), supported_frame_lengths_ms(),
+                  AudioEncoderOpusConfig::kMinBitrateBps,
+                  num_channels_to_encode_, next_frame_length_ms_,
+                  GetTargetBitrate(), config_.fec_enabled, GetDtx())));
 }
 
 void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {