Removing AudioCoding class, a.k.a the new ACM API

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

Index: webrtc/modules/audio_coding/main/interface/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index d0b7b03c43c1e228d4e25ea9d98c389de9b754e5..830f1e5b7710a9593ca2ac00ba4c3d86eaff3ff0 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -753,177 +753,6 @@ class AudioCodingModule {
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
-class AudioEncoder;
-class ReceiverInfo;
-
-class AudioCoding {
- public:
-  struct Config {
-    Config();
-
-    AudioCodingModule::Config ToOldConfig() const {
-      AudioCodingModule::Config old_config;
-      old_config.id = 0;
-      old_config.neteq_config = neteq_config;
-      old_config.clock = clock;
-      return old_config;
-    }
-
-    NetEq::Config neteq_config;
-    Clock* clock;
-    AudioPacketizationCallback* transport;
-    ACMVADCallback* vad_callback;
-    int initial_playout_delay_ms;
-    int playout_channels;
-    int playout_frequency_hz;
-  };
-
-  static AudioCoding* Create(const Config& config);
-  virtual ~AudioCoding() {};
-
-  // Registers a codec, specified by |send_codec|, as sending codec.
-  // This API can be called multiple times. The last codec registered overwrites
-  // the previous ones. Returns true if successful, false if not.
-  //
-  // Note: If a stereo codec is registered as send codec, VAD/DTX will
-  // automatically be turned off, since it is not supported for stereo sending.
-  virtual bool RegisterSendCodec(AudioEncoder* send_codec) = 0;
-
-  // Temporary solution to be used during refactoring:
-  // |encoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
-  virtual bool RegisterSendCodec(int encoder_type,
-                                 uint8_t payload_type,
-                                 int frame_size_samples = 0) = 0;
-
-  // Returns the encoder object currently in use. This is the same as the
-  // codec that was registered in the latest call to RegisterSendCodec().
-  virtual const AudioEncoder* GetSenderInfo() const = 0;
-
-  // Temporary solution to be used during refactoring.
-  virtual const CodecInst* GetSenderCodecInst() = 0;
-
-  // Adds 10 ms of raw (PCM) audio data to the encoder. If the sampling
-  // frequency of the audio does not match the sampling frequency of the
-  // current encoder, ACM will resample the audio.
-  //
-  // Return value:
-  //      0   successfully added the frame.
-  //     -1   some error occurred and data is not added.
-  //   < -1   to add the frame to the buffer n samples had to be
-  //          overwritten, -n is the return value in this case.
-  // TODO(henrik.lundin): Make a better design for the return values. This one
-  // is just a copy of the old API.
-  virtual int Add10MsAudio(const AudioFrame& audio_frame) = 0;
-
-  // Returns a combined info about the currently used decoder(s).
-  virtual const ReceiverInfo* GetReceiverInfo() const = 0;
-
-  // Registers a codec, specified by |receive_codec|, as receiving codec.
-  // This API can be called multiple times. If registering with a payload type
-  // that was already registered in a previous call, the latest call will
-  // override previous calls. Returns true if successful, false if not.
-  virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) = 0;
-
-  // Temporary solution:
-  // |decoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
-  virtual bool RegisterReceiveCodec(int decoder_type, uint8_t payload_type) = 0;
-
-  // The following two methods both inserts a new packet to the receiver.
-  // InsertPacket takes an RTP header input in |rtp_info|, while InsertPayload
-  // only requires a payload type and a timestamp. The latter assumes that the
-  // payloads come in the right order, and without any losses. In both cases,
-  // |incoming_payload| contains the RTP payload after the RTP header. Return
-  // true if successful, false if not.
-  virtual bool InsertPacket(const uint8_t* incoming_payload,
-                            size_t payload_len_bytes,
-                            const WebRtcRTPHeader& rtp_info) = 0;
-
-  // TODO(henrik.lundin): Remove this method?
-  virtual bool InsertPayload(const uint8_t* incoming_payload,
-                             size_t payload_len_byte,
-                             uint8_t payload_type,
-                             uint32_t timestamp) = 0;
-
-  // These two methods set a minimum and maximum jitter buffer delay in
-  // milliseconds. The pupose is mainly to adjust the delay to synchronize
-  // audio and video. The preferred jitter buffer size, computed by NetEq based
-  // on the current channel conditions, is clamped from below and above by these
-  // two methods. The given delay limits must be non-negative, less than
-  // 10000 ms, and the minimum must be strictly smaller than the maximum.
-  // Further, the maximum must be at lest one frame duration. If these
-  // conditions are not met, false is returned. Giving the value 0 effectively
-  // unsets the minimum or maximum delay limits.
-  // Note that calling these methods is optional. If not called, NetEq will
-  // determine the optimal buffer size based on the network conditions.
-  virtual bool SetMinimumPlayoutDelay(int time_ms) = 0;
-
-  virtual bool SetMaximumPlayoutDelay(int time_ms) = 0;
-
-  // Returns the current value of the jitter buffer's preferred latency. This
-  // is computed based on inter-arrival times and playout mode of NetEq. The
-  // actual target delay is this value clamped from below and above by the
-  // values specified through SetMinimumPlayoutDelay() and
-  // SetMaximumPlayoutDelay(), respectively, if provided.
-  // TODO(henrik.lundin) Rename to PreferredDelayMs?
-  virtual int LeastRequiredDelayMs() const = 0;
-
-  // The send timestamp of an RTP packet is associated with the decoded
-  // audio of the packet in question. This function returns the timestamp of
-  // the latest audio delivered by Get10MsAudio(). Returns false if no timestamp
-  // can be provided, true otherwise.
-  virtual bool PlayoutTimestamp(uint32_t* timestamp) = 0;
-
-  // Delivers 10 ms of audio in |audio_frame|. Returns true if successful,
-  // false otherwise.
-  virtual bool Get10MsAudio(AudioFrame* audio_frame) = 0;
-
-  // Returns the network statistics. Note that the internal statistics of NetEq
-  // are reset by this call. Returns true if successful, false otherwise.
-  virtual bool GetNetworkStatistics(NetworkStatistics* network_statistics) = 0;
-
-  // Enables NACK and sets the maximum size of the NACK list. If NACK is already
-  // enabled then the maximum NACK list size is modified accordingly. Returns
-  // true if successful, false otherwise.
-  //
-  // If the sequence number of last received packet is N, the sequence numbers
-  // of NACK list are in the range of [N - |max_nack_list_size|, N).
-  //
-  // |max_nack_list_size| should be positive and less than or equal to
-  // |Nack::kNackListSizeLimit|.
-  virtual bool EnableNack(size_t max_nack_list_size) = 0;
-
-  // Disables NACK.
-  virtual void DisableNack() = 0;
-
-
-  // Temporary solution to be used during refactoring.
-  // If DTX is enabled and the codec does not have internal DTX/VAD
-  // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
-  //
-  // If DTX is disabled but VAD is enabled no DTX packets are sent,
-  // regardless of whether the codec has internal DTX/VAD or not. In this
-  // case, WebRtc VAD is running to label frames as active/in-active.
-  //
-  // NOTE! VAD/DTX is not supported when sending stereo.
-  //
-  // Return true if successful, false otherwise.
-  virtual bool SetVad(bool enable_dtx,
-                      bool enable_vad,
-                      ACMVADMode vad_mode) = 0;
-
-  // Returns a list of packets to request retransmission of.
-  // |round_trip_time_ms| is an estimate of the round-trip-time (in
-  // milliseconds). Missing packets which will be decoded sooner than the
-  // round-trip-time (with respect to the time this API is called) will not be
-  // included in the list.
-  // |round_trip_time_ms| must be non-negative.
-  virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
-
-  // Returns the timing statistics for calls to Get10MsAudio.
-  virtual void GetDecodingCallStatistics(
-      AudioDecodingCallStats* call_stats) const = 0;
-};
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_