Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
deleted file mode 100644
index c098e62b9914da91662e7690a5b73c8e61f8a316..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- *  Copyright (c) 2012 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
-
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-
-namespace webrtc {
-
-class AudioCodingImpl;
-
-namespace acm2 {
-
-struct EncoderFactory;
-
-class AudioCodingModuleImpl final : public AudioCodingModule {
- public:
-  friend webrtc::AudioCodingImpl;
-
-  explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
-  ~AudioCodingModuleImpl() override;
-
-  /////////////////////////////////////////
-  //   Sender
-  //
-
-  // Can be called multiple times for Codec, CNG, RED.
-  int RegisterSendCodec(const CodecInst& send_codec) override;
-
-  void RegisterExternalSendCodec(
-      AudioEncoder* external_speech_encoder) override;
-
-  void ModifyEncoder(
-      FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) override;
-
-  // Get current send codec.
-  rtc::Optional<CodecInst> SendCodec() const override;
-
-  // Get current send frequency.
-  int SendFrequency() const override;
-
-  // Sets the bitrate to the specified value in bits/sec. In case the codec does
-  // not support the requested value it will choose an appropriate value
-  // instead.
-  void SetBitRate(int bitrate_bps) override;
-
-  // Register a transport callback which will be
-  // called to deliver the encoded buffers.
-  int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
-
-  // Add 10 ms of raw (PCM) audio data to the encoder.
-  int Add10MsData(const AudioFrame& audio_frame) override;
-
-  /////////////////////////////////////////
-  // (RED) Redundant Coding
-  //
-
-  // Configure RED status i.e. on/off.
-  int SetREDStatus(bool enable_red) override;
-
-  // Get RED status.
-  bool REDStatus() const override;
-
-  /////////////////////////////////////////
-  // (FEC) Forward Error Correction (codec internal)
-  //
-
-  // Configure FEC status i.e. on/off.
-  int SetCodecFEC(bool enabled_codec_fec) override;
-
-  // Get FEC status.
-  bool CodecFEC() const override;
-
-  // Set target packet loss rate
-  int SetPacketLossRate(int loss_rate) override;
-
-  /////////////////////////////////////////
-  //   (VAD) Voice Activity Detection
-  //   and
-  //   (CNG) Comfort Noise Generation
-  //
-
-  int SetVAD(bool enable_dtx = true,
-             bool enable_vad = false,
-             ACMVADMode mode = VADNormal) override;
-
-  int VAD(bool* dtx_enabled,
-          bool* vad_enabled,
-          ACMVADMode* mode) const override;
-
-  int RegisterVADCallback(ACMVADCallback* vad_callback) override;
-
-  /////////////////////////////////////////
-  //   Receiver
-  //
-
-  // Initialize receiver, resets codec database etc.
-  int InitializeReceiver() override;
-
-  // Get current receive frequency.
-  int ReceiveFrequency() const override;
-
-  // Get current playout frequency.
-  int PlayoutFrequency() const override;
-
-  int RegisterReceiveCodec(const CodecInst& receive_codec) override;
-  int RegisterReceiveCodec(
-      const CodecInst& receive_codec,
-      FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) override;
-
-  int RegisterExternalReceiveCodec(int rtp_payload_type,
-                                   AudioDecoder* external_decoder,
-                                   int sample_rate_hz,
-                                   int num_channels,
-                                   const std::string& name) override;
-
-  // Get current received codec.
-  int ReceiveCodec(CodecInst* current_codec) const override;
-
-  // Incoming packet from network parsed and ready for decode.
-  int IncomingPacket(const uint8_t* incoming_payload,
-                     const size_t payload_length,
-                     const WebRtcRTPHeader& rtp_info) override;
-
-  // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
-  // One usage for this API is when pre-encoded files are pushed in ACM.
-  int IncomingPayload(const uint8_t* incoming_payload,
-                      const size_t payload_length,
-                      uint8_t payload_type,
-                      uint32_t timestamp) override;
-
-  // Minimum playout delay.
-  int SetMinimumPlayoutDelay(int time_ms) override;
-
-  // Maximum playout delay.
-  int SetMaximumPlayoutDelay(int time_ms) override;
-
-  // Smallest latency NetEq will maintain.
-  int LeastRequiredDelayMs() const override;
-
-  RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
-
-  rtc::Optional<uint32_t> PlayoutTimestamp() override;
-
-  // Get 10 milliseconds of raw audio data to play out, and
-  // automatic resample to the requested frequency if > 0.
-  int PlayoutData10Ms(int desired_freq_hz,
-                      AudioFrame* audio_frame,
-                      bool* muted) override;
-  int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
-
-  /////////////////////////////////////////
-  //   Statistics
-  //
-
-  int GetNetworkStatistics(NetworkStatistics* statistics) override;
-
-  int SetOpusApplication(OpusApplicationMode application) override;
-
-  // If current send codec is Opus, informs it about the maximum playback rate
-  // the receiver will render.
-  int SetOpusMaxPlaybackRate(int frequency_hz) override;
-
-  int EnableOpusDtx() override;
-
-  int DisableOpusDtx() override;
-
-  int UnregisterReceiveCodec(uint8_t payload_type) override;
-
-  int EnableNack(size_t max_nack_list_size) override;
-
-  void DisableNack() override;
-
-  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
-
-  void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
-
- private:
-  struct InputData {
-    uint32_t input_timestamp;
-    const int16_t* audio;
-    size_t length_per_channel;
-    size_t audio_channel;
-    // If a re-mix is required (up or down), this buffer will store a re-mixed
-    // version of the input.
-    int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
-  };
-
-  // This member class writes values to the named UMA histogram, but only if
-  // the value has changed since the last time (and always for the first call).
-  class ChangeLogger {
-   public:
-    explicit ChangeLogger(const std::string& histogram_name)
-        : histogram_name_(histogram_name) {}
-    // Logs the new value if it is different from the last logged value, or if
-    // this is the first call.
-    void MaybeLog(int value);
-
-   private:
-    int last_value_ = 0;
-    int first_time_ = true;
-    const std::string histogram_name_;
-  };
-
-  int RegisterReceiveCodecUnlocked(
-      const CodecInst& codec,
-      FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory)
-      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-
-  int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
-      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-  int Encode(const InputData& input_data)
-      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-
-  int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-
-  bool HaveValidEncoder(const char* caller_name) const
-      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-
-  // Preprocessing of input audio, including resampling and down-mixing if
-  // required, before pushing audio into encoder's buffer.
-  //
-  // in_frame: input audio-frame
-  // ptr_out: pointer to output audio_frame. If no preprocessing is required
-  //          |ptr_out| will be pointing to |in_frame|, otherwise pointing to
-  //          |preprocess_frame_|.
-  //
-  // Return value:
-  //   -1: if encountering an error.
-  //    0: otherwise.
-  int PreprocessToAddData(const AudioFrame& in_frame,
-                          const AudioFrame** ptr_out)
-      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-
-  // Change required states after starting to receive the codec corresponding
-  // to |index|.
-  int UpdateUponReceivingCodec(int index);
-
-  rtc::CriticalSection acm_crit_sect_;
-  rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
-  int id_;  // TODO(henrik.lundin) Make const.
-  uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
-  uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
-  ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
-  AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
-  ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
-
-  std::unique_ptr<EncoderFactory> encoder_factory_ GUARDED_BY(acm_crit_sect_);
-
-  // Current encoder stack, either obtained from
-  // encoder_factory_->rent_a_codec.RentEncoderStack or provided by a call to
-  // RegisterEncoder.
-  std::unique_ptr<AudioEncoder> encoder_stack_ GUARDED_BY(acm_crit_sect_);
-
-  std::unique_ptr<AudioDecoder> isac_decoder_16k_ GUARDED_BY(acm_crit_sect_);
-  std::unique_ptr<AudioDecoder> isac_decoder_32k_ GUARDED_BY(acm_crit_sect_);
-
-  // This is to keep track of CN instances where we can send DTMFs.
-  uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
-
-  // Used when payloads are pushed into ACM without any RTP info
-  // One example is when pre-encoded bit-stream is pushed from
-  // a file.
-  // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
-  // no lock acquired when interacting with this variable. If it is going to
-  // be used in other methods, locks need to be taken.
-  std::unique_ptr<WebRtcRTPHeader> aux_rtp_header_;
-
-  bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
-
-  AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
-  bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
-
-  bool first_frame_ GUARDED_BY(acm_crit_sect_);
-  uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
-  uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
-
-  rtc::CriticalSection callback_crit_sect_;
-  AudioPacketizationCallback* packetization_callback_
-      GUARDED_BY(callback_crit_sect_);
-  ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
-
-  int codec_histogram_bins_log_[static_cast<size_t>(
-      AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
-  int number_of_consecutive_empty_packets_;
-};
-
-}  // namespace acm2
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_