Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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/g711/include/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
+
 int16_t NumSamplesPerFrame(int num_channels,
                            int frame_size_ms,
                            int sample_rate_hz) {
   int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
   CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
       << "Frame size too large.";
   return static_cast<int16_t>(samples_per_frame);
 }
+
+template <typename T>
+typename T::Config CreateConfig(const CodecInst& codec_inst) {
+  typename T::Config config;
+  config.frame_size_ms = codec_inst.pacsize / 8;
+  config.num_channels = codec_inst.channels;
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
+
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       full_frame_samples_(NumSamplesPerFrame(config.num_channels,
                                              config.frame_size_ms,
                                              sample_rate_hz_)),
       first_timestamp_in_buffer_(0) {
   CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
   CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   speech_buffer_.reserve(full_frame_samples_);
 }
 
-AudioEncoderPcm::~AudioEncoderPcm() {
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+size_t AudioEncoderPcm::MaxEncodedBytes() const {
+  return full_frame_samples_ * BytesPerSample();
 }
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
 int AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
-size_t AudioEncoderPcm::MaxEncodedBytes() const {
-  return full_frame_samples_ * BytesPerSample();
-}
-
 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
   return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
@@ skipping 18 matching lines @@
   CHECK_GE(max_encoded_bytes, full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes =
       EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
   speech_buffer_.clear();
   return info;
 }
 
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
+    : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
+
 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
 int AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
+AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
+    : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
+
 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 int AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
-namespace {
-template <typename T>
-typename T::Config CreateConfig(const CodecInst& codec_inst) {
-  typename T::Config config;
-  config.frame_size_ms = codec_inst.pacsize / 8;
-  config.num_channels = codec_inst.channels;
-  config.payload_type = codec_inst.pltype;
-  return config;
-}
-}  // namespace
-
-AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst)
-    : AudioEncoderMutableImpl<AudioEncoderPcmU>(
-          CreateConfig<AudioEncoderPcmU>(codec_inst)) {
-}
-
-AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst)
-    : AudioEncoderMutableImpl<AudioEncoderPcmA>(
-          CreateConfig<AudioEncoderPcmA>(codec_inst)) {
-}
-
 }  // namespace webrtc