Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
diff --git a/webrtc/api/audio_codecs/audio_encoder.h b/webrtc/api/audio_codecs/audio_encoder.h
new file mode 100644
index 0000000..3e81000
--- /dev/null
+++ b/webrtc/api/audio_codecs/audio_encoder.h
@@ -0,0 +1,214 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
+#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#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);
+
+ 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_API_AUDIO_CODECS_AUDIO_ENCODER_H_
