Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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182
modules/audio_coding/codecs/opus/audio_encoder_opus.h
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182
modules/audio_coding/codecs/opus/audio_encoder_opus.h
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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#include <functional>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_encoder.h"
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#include "webrtc/api/audio_codecs/audio_format.h"
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#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
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#include "webrtc/api/optional.h"
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#include "webrtc/common_audio/smoothing_filter.h"
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#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
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#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/protobuf_utils.h"
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namespace webrtc {
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class RtcEventLog;
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struct CodecInst;
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class AudioEncoderOpus final : public AudioEncoder {
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public:
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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const AudioEncoderOpusConfig&,
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int payload_type);
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// NOTE: This alias will soon go away. See
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
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using Config = AudioEncoderOpusConfig;
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// NOTE: This function will soon go away. See
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
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static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
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static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
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static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
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const SdpAudioFormat& format);
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// Returns empty if the current bitrate falls within the hysteresis window,
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// defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
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// Otherwise, returns the current complexity depending on whether the
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// current bitrate is above or below complexity_threshold_bps.
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static rtc::Optional<int> GetNewComplexity(
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const AudioEncoderOpusConfig& config);
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using AudioNetworkAdaptorCreator =
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std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
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RtcEventLog*)>;
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// NOTE: This constructor will soon go away. See
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
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AudioEncoderOpus(const AudioEncoderOpusConfig& config);
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AudioEncoderOpus(const AudioEncoderOpusConfig& config, int payload_type);
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// Dependency injection for testing.
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AudioEncoderOpus(
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const AudioEncoderOpusConfig& config,
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int payload_type,
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const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
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std::unique_ptr<SmoothingFilter> bitrate_smoother);
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explicit AudioEncoderOpus(const CodecInst& codec_inst);
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AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
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~AudioEncoderOpus() override;
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// Static interface for use by BuiltinAudioEncoderFactory.
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static constexpr const char* GetPayloadName() { return "opus"; }
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static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
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const SdpAudioFormat& format);
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int SampleRateHz() const override;
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size_t NumChannels() const override;
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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int GetTargetBitrate() const override;
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void Reset() override;
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bool SetFec(bool enable) override;
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// Set Opus DTX. Once enabled, Opus stops transmission, when it detects
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// voice being inactive. During that, it still sends 2 packets (one for
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// content, one for signaling) about every 400 ms.
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bool SetDtx(bool enable) override;
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bool GetDtx() const override;
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bool SetApplication(Application application) override;
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void SetMaxPlaybackRate(int frequency_hz) override;
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bool EnableAudioNetworkAdaptor(const std::string& config_string,
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RtcEventLog* event_log) override;
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void DisableAudioNetworkAdaptor() override;
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void OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) override;
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void OnReceivedUplinkRecoverablePacketLossFraction(
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float uplink_recoverable_packet_loss_fraction) override;
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void OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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rtc::Optional<int64_t> bwe_period_ms) override;
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void OnReceivedRtt(int rtt_ms) override;
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void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
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void SetReceiverFrameLengthRange(int min_frame_length_ms,
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int max_frame_length_ms) override;
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ANAStats GetANAStats() const override;
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rtc::ArrayView<const int> supported_frame_lengths_ms() const {
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return config_.supported_frame_lengths_ms;
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}
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// Getters for testing.
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float packet_loss_rate() const { return packet_loss_rate_; }
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AudioEncoderOpusConfig::ApplicationMode application() const {
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return config_.application;
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}
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bool fec_enabled() const { return config_.fec_enabled; }
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size_t num_channels_to_encode() const { return num_channels_to_encode_; }
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int next_frame_length_ms() const { return next_frame_length_ms_; }
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protected:
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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private:
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class PacketLossFractionSmoother;
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size_t Num10msFramesPerPacket() const;
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size_t SamplesPer10msFrame() const;
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size_t SufficientOutputBufferSize() const;
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bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
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void SetFrameLength(int frame_length_ms);
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void SetNumChannelsToEncode(size_t num_channels_to_encode);
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void SetProjectedPacketLossRate(float fraction);
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// TODO(minyue): remove "override" when we can deprecate
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// |AudioEncoder::SetTargetBitrate|.
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void SetTargetBitrate(int target_bps) override;
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void ApplyAudioNetworkAdaptor();
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std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
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const ProtoString& config_string,
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RtcEventLog* event_log) const;
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void MaybeUpdateUplinkBandwidth();
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AudioEncoderOpusConfig config_;
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const int payload_type_;
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const bool send_side_bwe_with_overhead_;
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float packet_loss_rate_;
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std::vector<int16_t> input_buffer_;
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OpusEncInst* inst_;
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uint32_t first_timestamp_in_buffer_;
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size_t num_channels_to_encode_;
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int next_frame_length_ms_;
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int complexity_;
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std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
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const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
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std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
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rtc::Optional<size_t> overhead_bytes_per_packet_;
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const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
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rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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