Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index bdbbf2d45b7871cf1ab2eaaf3327e63612d93153..942abb6f0dc8a59f95a7f909335cbd2ffa4c37bb 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -8,205 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_encoder.h instead!
+// TODO(ossu): Remove it.
+
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
-#include <algorithm>
-#include <vector>
-
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/deprecation.h"
-#include "webrtc/base/optional.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class Clock;
-class RtcEventLog;
-
-// This is the interface class for encoders in AudioCoding module. Each codec
-// type must have an implementation of this class.
-class AudioEncoder {
- public:
-  // Used for UMA logging of codec usage. The same codecs, with the
-  // same values, must be listed in
-  // src/tools/metrics/histograms/histograms.xml in chromium to log
-  // correct values.
-  enum class CodecType {
-    kOther = 0,  // Codec not specified, and/or not listed in this enum
-    kOpus = 1,
-    kIsac = 2,
-    kPcmA = 3,
-    kPcmU = 4,
-    kG722 = 5,
-    kIlbc = 6,
-
-    // Number of histogram bins in the UMA logging of codec types. The
-    // total number of different codecs that are logged cannot exceed this
-    // number.
-    kMaxLoggedAudioCodecTypes
-  };
-
-  struct EncodedInfoLeaf {
-    size_t encoded_bytes = 0;
-    uint32_t encoded_timestamp = 0;
-    int payload_type = 0;
-    bool send_even_if_empty = false;
-    bool speech = true;
-    CodecType encoder_type = CodecType::kOther;
-  };
-
-  // This is the main struct for auxiliary encoding information. Each encoded
-  // packet should be accompanied by one EncodedInfo struct, containing the
-  // total number of |encoded_bytes|, the |encoded_timestamp| and the
-  // |payload_type|. If the packet contains redundant encodings, the |redundant|
-  // vector will be populated with EncodedInfoLeaf structs. Each struct in the
-  // vector represents one encoding; the order of structs in the vector is the
-  // same as the order in which the actual payloads are written to the byte
-  // stream. When EncoderInfoLeaf structs are present in the vector, the main
-  // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
-  // vector.
-  struct EncodedInfo : public EncodedInfoLeaf {
-    EncodedInfo();
-    EncodedInfo(const EncodedInfo&);
-    EncodedInfo(EncodedInfo&&);
-    ~EncodedInfo();
-    EncodedInfo& operator=(const EncodedInfo&);
-    EncodedInfo& operator=(EncodedInfo&&);
-
-    std::vector<EncodedInfoLeaf> redundant;
-  };
-
-  virtual ~AudioEncoder() = default;
-
-  // Returns the input sample rate in Hz and the number of input channels.
-  // These are constants set at instantiation time.
-  virtual int SampleRateHz() const = 0;
-  virtual size_t NumChannels() const = 0;
-
-  // Returns the rate at which the RTP timestamps are updated. The default
-  // implementation returns SampleRateHz().
-  virtual int RtpTimestampRateHz() const;
-
-  // Returns the number of 10 ms frames the encoder will put in the next
-  // packet. This value may only change when Encode() outputs a packet; i.e.,
-  // the encoder may vary the number of 10 ms frames from packet to packet, but
-  // it must decide the length of the next packet no later than when outputting
-  // the preceding packet.
-  virtual size_t Num10MsFramesInNextPacket() const = 0;
-
-  // Returns the maximum value that can be returned by
-  // Num10MsFramesInNextPacket().
-  virtual size_t Max10MsFramesInAPacket() const = 0;
-
-  // Returns the current target bitrate in bits/s. The value -1 means that the
-  // codec adapts the target automatically, and a current target cannot be
-  // provided.
-  virtual int GetTargetBitrate() const = 0;
-
-  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
-  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
-  // The encoder appends zero or more bytes of output to |encoded| and returns
-  // additional encoding information.  Encode() checks some preconditions, calls
-  // EncodeImpl() which does the actual work, and then checks some
-  // postconditions.
-  EncodedInfo Encode(uint32_t rtp_timestamp,
-                     rtc::ArrayView<const int16_t> audio,
-                     rtc::Buffer* encoded);
-
-  // Resets the encoder to its starting state, discarding any input that has
-  // been fed to the encoder but not yet emitted in a packet.
-  virtual void Reset() = 0;
-
-  // Enables or disables codec-internal FEC (forward error correction). Returns
-  // true if the codec was able to comply. The default implementation returns
-  // true when asked to disable FEC and false when asked to enable it (meaning
-  // that FEC isn't supported).
-  virtual bool SetFec(bool enable);
-
-  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
-  // able to comply. The default implementation returns true when asked to
-  // disable DTX and false when asked to enable it (meaning that DTX isn't
-  // supported).
-  virtual bool SetDtx(bool enable);
-
-  // Returns the status of codec-internal DTX. The default implementation always
-  // returns false.
-  virtual bool GetDtx() const;
-
-  // Sets the application mode. Returns true if the codec was able to comply.
-  // The default implementation just returns false.
-  enum class Application { kSpeech, kAudio };
-  virtual bool SetApplication(Application application);
-
-  // Tells the encoder about the highest sample rate the decoder is expected to
-  // use when decoding the bitstream. The encoder would typically use this
-  // information to adjust the quality of the encoding. The default
-  // implementation does nothing.
-  virtual void SetMaxPlaybackRate(int frequency_hz);
-
-  // This is to be deprecated. Please use |OnReceivedTargetAudioBitrate|
-  // instead.
-  // Tells the encoder what average bitrate we'd like it to produce. The
-  // encoder is free to adjust or disregard the given bitrate (the default
-  // implementation does the latter).
-  RTC_DEPRECATED virtual void SetTargetBitrate(int target_bps);
-
-  // Causes this encoder to let go of any other encoders it contains, and
-  // returns a pointer to an array where they are stored (which is required to
-  // live as long as this encoder). Unless the returned array is empty, you may
-  // not call any methods on this encoder afterwards, except for the
-  // destructor. The default implementation just returns an empty array.
-  // NOTE: This method is subject to change. Do not call or override it.
-  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
-  ReclaimContainedEncoders();
-
-  // Enables audio network adaptor. Returns true if successful.
-  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
-                                         RtcEventLog* event_log);
-
-  // Disables audio network adaptor.
-  virtual void DisableAudioNetworkAdaptor();
-
-  // Provides uplink packet loss fraction to this encoder to allow it to adapt.
-  // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
-  virtual void OnReceivedUplinkPacketLossFraction(
-      float uplink_packet_loss_fraction);
-
-  // Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
-  // to allow it to adapt.
-  // |uplink_recoverable_packet_loss_fraction| is in the range [0.0, 1.0].
-  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
-      float uplink_recoverable_packet_loss_fraction);
-
-  // Provides target audio bitrate to this encoder to allow it to adapt.
-  virtual void OnReceivedTargetAudioBitrate(int target_bps);
-
-  // Provides target audio bitrate and corresponding probing interval of
-  // the bandwidth estimator to this encoder to allow it to adapt.
-  virtual void OnReceivedUplinkBandwidth(
-      int target_audio_bitrate_bps,
-      rtc::Optional<int64_t> probing_interval_ms);
-
-  // Provides RTT to this encoder to allow it to adapt.
-  virtual void OnReceivedRtt(int rtt_ms);
-
-  // Provides overhead to this encoder to adapt. The overhead is the number of
-  // bytes that will be added to each packet the encoder generates.
-  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
-
-  // To allow encoder to adapt its frame length, it must be provided the frame
-  // length range that receivers can accept.
-  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
-                                           int max_frame_length_ms);
+#include "webrtc/api/audio_codecs/audio_encoder.h"
 
- protected:
-  // Subclasses implement this to perform the actual encoding. Called by
-  // Encode().
-  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
-                                 rtc::ArrayView<const int16_t> audio,
-                                 rtc::Buffer* encoded) = 0;
-};
-}  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_