Fold AudioEncoderMutable into AudioEncoder

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 37ce8733fed21268db49a942eb3cb1c7a7a1b76b..e0f59adaadcd010ccff709e7a9f56e8641971fee 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
 
@@ -18,36 +19,61 @@ namespace webrtc {
 
 namespace {
 
+const int kSampleRateHz = 48000;
 const int kMinBitrateBps = 500;
 const int kMaxBitrateBps = 512000;
 
-// TODO(tlegrand): Remove this code when we have proper APIs to set the
-// complexity at a higher level.
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
-// If we are on Android, iOS and/or ARM, use a lower complexity setting as
-// default, to save encoder complexity.
-const int kDefaultComplexity = 5;
-#else
-const int kDefaultComplexity = 9;
-#endif
+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.payload_type = codec_inst.pltype;
+  config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
+                                                : AudioEncoderOpus::kAudio;
+  return config;
+}
 
-// We always encode at 48 kHz.
-const int kSampleRateHz = 48000;
+// Optimize the loss rate to configure Opus. Basically, optimized loss rate is
+// the input loss rate rounded down to various levels, because a robustly good
+// audio quality is achieved by lowering the packet loss down.
+// Additionally, to prevent toggling, margins are used, i.e., when jumping to
+// a loss rate from below, a higher threshold is used than jumping to the same
+// level from above.
+double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
+  DCHECK_GE(new_loss_rate, 0.0);
+  DCHECK_LE(new_loss_rate, 1.0);

[hlundin-webrtc, 2015/09/07 20:00:01]

DCHECKs for old_loss_rate too?

[kwiberg-webrtc, 2015/09/08 10:47:45]

Good idea. Done.

+  const double kPacketLossRate20 = 0.20;
+  const double kPacketLossRate10 = 0.10;
+  const double kPacketLossRate5 = 0.05;
+  const double kPacketLossRate1 = 0.01;
+  const double kLossRate20Margin = 0.02;
+  const double kLossRate10Margin = 0.01;
+  const double kLossRate5Margin = 0.01;
+  if (new_loss_rate >=
+      kPacketLossRate20 +
+          kLossRate20Margin *
+              (kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate20;
+  } else if (new_loss_rate >=
+             kPacketLossRate10 +
+                 kLossRate10Margin *
+                     (kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate10;
+  } else if (new_loss_rate >=
+             kPacketLossRate5 +
+                 kLossRate5Margin *
+                     (kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate5;
+  } else if (new_loss_rate >= kPacketLossRate1) {
+    return kPacketLossRate1;
+  } else {
+    return 0.0;
+  }
+}
 
 }  // namespace
 
-AudioEncoderOpus::Config::Config()
-    : frame_size_ms(20),
-      num_channels(1),
-      payload_type(120),
-      application(kVoip),
-      bitrate_bps(64000),
-      fec_enabled(false),
-      max_playback_rate_hz(48000),
-      complexity(kDefaultComplexity),
-      dtx_enabled(false) {
-}
-
 bool AudioEncoderOpus::Config::IsOk() const {
   if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
     return false;
@@ -61,119 +87,45 @@ bool AudioEncoderOpus::Config::IsOk() const {
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const Config& config)
-    : num_10ms_frames_per_packet_(
-          static_cast<size_t>(rtc::CheckedDivExact(config.frame_size_ms, 10))),
-      num_channels_(config.num_channels),
-      payload_type_(config.payload_type),
-      application_(config.application),
-      dtx_enabled_(config.dtx_enabled),
-      samples_per_10ms_frame_(static_cast<size_t>(
-          rtc::CheckedDivExact(kSampleRateHz, 100) * num_channels_)),
-      packet_loss_rate_(0.0) {
-  CHECK(config.IsOk());
-  input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
-  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
-  SetTargetBitrate(config.bitrate_bps);
-  if (config.fec_enabled) {
-    CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
-  } else {
-    CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
-  }
-  CHECK_EQ(0,
-           WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
-  CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
-  if (config.dtx_enabled) {
-    CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
-  } else {
-    CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
-  }
+    : packet_loss_rate_(0.0), inst_(nullptr) {
+  CHECK(RecreateEncoderInstance(config));
 }
 
+AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
+    : AudioEncoderOpus(CreateConfig(codec_inst)) {}
+
 AudioEncoderOpus::~AudioEncoderOpus() {
   CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
-int AudioEncoderOpus::SampleRateHz() const {
-  return kSampleRateHz;
-}
-
-int AudioEncoderOpus::NumChannels() const {
-  return num_channels_;
-}
-
 size_t AudioEncoderOpus::MaxEncodedBytes() const {
   // Calculate the number of bytes we expect the encoder to produce,
   // then multiply by two to give a wide margin for error.
-  size_t bytes_per_millisecond =
-      static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
-  size_t approx_encoded_bytes =
-      num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond;
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
-size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const {
-  return num_10ms_frames_per_packet_;
+int AudioEncoderOpus::SampleRateHz() const {
+  return kSampleRateHz;
 }
 
-size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
-  return num_10ms_frames_per_packet_;
+int AudioEncoderOpus::NumChannels() const {
+  return config_.num_channels;
 }
 
-int AudioEncoderOpus::GetTargetBitrate() const {
-  return bitrate_bps_;
+size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
 }
 
-void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
-  bitrate_bps_ = std::max(std::min(bits_per_second, kMaxBitrateBps),
-                          kMinBitrateBps);
-  CHECK_EQ(WebRtcOpus_SetBitRate(inst_, bitrate_bps_), 0);
+size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
 }
 
-void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
-  DCHECK_GE(fraction, 0.0);
-  DCHECK_LE(fraction, 1.0);
-  // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
-  // the input loss rate rounded down to various levels, because a robustly good
-  // audio quality is achieved by lowering the packet loss down.
-  // Additionally, to prevent toggling, margins are used, i.e., when jumping to
-  // a loss rate from below, a higher threshold is used than jumping to the same
-  // level from above.
-  const double kPacketLossRate20 = 0.20;
-  const double kPacketLossRate10 = 0.10;
-  const double kPacketLossRate5 = 0.05;
-  const double kPacketLossRate1 = 0.01;
-  const double kLossRate20Margin = 0.02;
-  const double kLossRate10Margin = 0.01;
-  const double kLossRate5Margin = 0.01;
-  double opt_loss_rate;
-  if (fraction >=
-      kPacketLossRate20 +
-          kLossRate20Margin *
-              (kPacketLossRate20 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate20;
-  } else if (fraction >=
-             kPacketLossRate10 +
-                 kLossRate10Margin *
-                     (kPacketLossRate10 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate10;
-  } else if (fraction >=
-             kPacketLossRate5 +
-                 kLossRate5Margin *
-                     (kPacketLossRate5 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate5;
-  } else if (fraction >= kPacketLossRate1) {
-    opt_loss_rate = kPacketLossRate1;
-  } else {
-    opt_loss_rate = 0;
-  }
-
-  if (packet_loss_rate_ != opt_loss_rate) {
-    // Ask the encoder to change the target packet loss rate.
-    CHECK_EQ(WebRtcOpus_SetPacketLossRate(
-                 inst_, static_cast<int32_t>(opt_loss_rate * 100 + .5)),
-             0);
-    packet_loss_rate_ = opt_loss_rate;
-  }
+int AudioEncoderOpus::GetTargetBitrate() const {
+  return config_.bitrate_bps;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
@@ -184,75 +136,118 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
   input_buffer_.insert(input_buffer_.end(), audio,
-                       audio + samples_per_10ms_frame_);
+                       audio + SamplesPer10msFrame());
   if (input_buffer_.size() <
-      (num_10ms_frames_per_packet_ * samples_per_10ms_frame_)) {
+      (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
-  CHECK_EQ(input_buffer_.size(),
-           num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
+  CHECK_EQ(input_buffer_.size(), static_cast<size_t>(Num10msFramesPerPacket()) *
+                                     SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(input_buffer_.size(),
-                           static_cast<size_t>(num_channels_)),
-      max_encoded_bytes, encoded);
+                           static_cast<size_t>(config_.num_channels)),
+      rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
   CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
   info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
-  info.payload_type = payload_type_;
+  info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (status > 0);
   return info;
 }
 
-namespace {
-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.payload_type = codec_inst.pltype;
-  config.application = (config.num_channels == 1 ? AudioEncoderOpus::kVoip
-                                                 : AudioEncoderOpus::kAudio);
-  return config;
+void AudioEncoderOpus::Reset() {
+  CHECK(RecreateEncoderInstance(config_));
 }
-}  // namespace
 
-AudioEncoderMutableOpus::AudioEncoderMutableOpus(const CodecInst& codec_inst)
-    : AudioEncoderMutableImpl<AudioEncoderOpus>(CreateConfig(codec_inst)) {
-}
-
-bool AudioEncoderMutableOpus::SetFec(bool enable) {
-  auto conf = config();
+bool AudioEncoderOpus::SetFec(bool enable) {
+  auto conf = config_;
   conf.fec_enabled = enable;
-  return Reconstruct(conf);
+  return RecreateEncoderInstance(conf);
 }
 
-bool AudioEncoderMutableOpus::SetDtx(bool enable) {
-  auto conf = config();
+bool AudioEncoderOpus::SetDtx(bool enable) {
+  auto conf = config_;
   conf.dtx_enabled = enable;
-  return Reconstruct(conf);
+  return RecreateEncoderInstance(conf);
 }
 
-bool AudioEncoderMutableOpus::SetApplication(Application application) {
-  auto conf = config();
+bool AudioEncoderOpus::SetApplication(Application application) {
+  auto conf = config_;
   switch (application) {
-    case kApplicationSpeech:
+    case Application::kSpeech:
       conf.application = AudioEncoderOpus::kVoip;
       break;
-    case kApplicationAudio:
+    case Application::kAudio:
       conf.application = AudioEncoderOpus::kAudio;
       break;
   }
-  return Reconstruct(conf);
+  return RecreateEncoderInstance(conf);
 }
 
-bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) {
-  auto conf = config();
+bool AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
+  auto conf = config_;
   conf.max_playback_rate_hz = frequency_hz;
-  return Reconstruct(conf);
+  return RecreateEncoderInstance(conf);
+}
+
+void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
+  double opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
+  if (packet_loss_rate_ != opt_loss_rate) {
+    packet_loss_rate_ = opt_loss_rate;
+    CHECK_EQ(0, WebRtcOpus_SetPacketLossRate(
+                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
+void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
+  config_.bitrate_bps =
+      std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps);
+  DCHECK(config_.IsOk());
+  CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
+}
+
+int AudioEncoderOpus::Num10msFramesPerPacket() const {
+  return rtc::CheckedDivExact(config_.frame_size_ms, 10);
+}
+
+int AudioEncoderOpus::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
+}
+
+// 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) {
+  if (!config.IsOk())
+    return false;
+  if (inst_)
+    CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
+                                       config.application));
+  CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.bitrate_bps));
+  if (config.fec_enabled) {
+    CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  CHECK_EQ(0,
+           WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
+  if (config.dtx_enabled) {
+    CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  CHECK_EQ(0, WebRtcOpus_SetPacketLossRate(
+                  inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  config_ = config;
+  return true;
 }
 
 }  // namespace webrtc