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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449
modules/audio_coding/neteq/neteq_impl.h
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449
modules/audio_coding/neteq/neteq_impl.h
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/*
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* Copyright (c) 2012 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_NETEQ_NETEQ_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
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#include <memory>
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#include <string>
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#include "webrtc/api/optional.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/defines.h"
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#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
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#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
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#include "webrtc/modules/audio_coding/neteq/random_vector.h"
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#include "webrtc/modules/audio_coding/neteq/rtcp.h"
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#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
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#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declarations.
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class Accelerate;
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class BackgroundNoise;
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class BufferLevelFilter;
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class ComfortNoise;
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class DecisionLogic;
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class DecoderDatabase;
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class DelayManager;
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class DelayPeakDetector;
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class DtmfBuffer;
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class DtmfToneGenerator;
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class Expand;
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class Merge;
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class NackTracker;
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class Normal;
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class PacketBuffer;
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class RedPayloadSplitter;
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class PostDecodeVad;
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class PreemptiveExpand;
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class RandomVector;
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class SyncBuffer;
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class TimestampScaler;
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struct AccelerateFactory;
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struct DtmfEvent;
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struct ExpandFactory;
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struct PreemptiveExpandFactory;
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class NetEqImpl : public webrtc::NetEq {
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public:
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enum class OutputType {
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kNormalSpeech,
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kPLC,
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kCNG,
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kPLCCNG,
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kVadPassive
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};
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enum ErrorCodes {
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kNoError = 0,
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kOtherError,
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kUnknownRtpPayloadType,
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kDecoderNotFound,
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kInvalidPointer,
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kAccelerateError,
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kPreemptiveExpandError,
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kComfortNoiseErrorCode,
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kDecoderErrorCode,
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kOtherDecoderError,
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kInvalidOperation,
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kDtmfParsingError,
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kDtmfInsertError,
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kSampleUnderrun,
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kDecodedTooMuch,
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kRedundancySplitError,
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kPacketBufferCorruption
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};
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struct Dependencies {
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// The constructor populates the Dependencies struct with the default
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// implementations of the objects. They can all be replaced by the user
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// before sending the struct to the NetEqImpl constructor. However, there
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// are dependencies between some of the classes inside the struct, so
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// swapping out one may make it necessary to re-create another one.
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explicit Dependencies(
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const NetEq::Config& config,
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const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
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~Dependencies();
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std::unique_ptr<TickTimer> tick_timer;
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std::unique_ptr<BufferLevelFilter> buffer_level_filter;
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std::unique_ptr<DecoderDatabase> decoder_database;
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std::unique_ptr<DelayPeakDetector> delay_peak_detector;
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std::unique_ptr<DelayManager> delay_manager;
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std::unique_ptr<DtmfBuffer> dtmf_buffer;
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std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
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std::unique_ptr<PacketBuffer> packet_buffer;
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std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
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std::unique_ptr<TimestampScaler> timestamp_scaler;
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std::unique_ptr<AccelerateFactory> accelerate_factory;
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std::unique_ptr<ExpandFactory> expand_factory;
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std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
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};
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// Creates a new NetEqImpl object.
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NetEqImpl(const NetEq::Config& config,
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Dependencies&& deps,
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bool create_components = true);
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~NetEqImpl() override;
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// Inserts a new packet into NetEq. The |receive_timestamp| is an indication
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// of the time when the packet was received, and should be measured with
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// the same tick rate as the RTP timestamp of the current payload.
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// Returns 0 on success, -1 on failure.
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int InsertPacket(const RTPHeader& rtp_header,
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rtc::ArrayView<const uint8_t> payload,
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uint32_t receive_timestamp) override;
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void InsertEmptyPacket(const RTPHeader& rtp_header) override;
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int GetAudio(AudioFrame* audio_frame, bool* muted) override;
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void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
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int RegisterPayloadType(NetEqDecoder codec,
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const std::string& codec_name,
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uint8_t rtp_payload_type) override;
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int RegisterExternalDecoder(AudioDecoder* decoder,
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NetEqDecoder codec,
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const std::string& codec_name,
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uint8_t rtp_payload_type) override;
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bool RegisterPayloadType(int rtp_payload_type,
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const SdpAudioFormat& audio_format) override;
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// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
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// -1 on failure.
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int RemovePayloadType(uint8_t rtp_payload_type) override;
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void RemoveAllPayloadTypes() override;
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bool SetMinimumDelay(int delay_ms) override;
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bool SetMaximumDelay(int delay_ms) override;
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int LeastRequiredDelayMs() const override;
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int SetTargetDelay() override;
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int TargetDelayMs() override;
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int CurrentDelayMs() const override;
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int FilteredCurrentDelayMs() const override;
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// Sets the playout mode to |mode|.
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// Deprecated.
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// TODO(henrik.lundin) Delete.
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void SetPlayoutMode(NetEqPlayoutMode mode) override;
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// Returns the current playout mode.
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// Deprecated.
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// TODO(henrik.lundin) Delete.
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NetEqPlayoutMode PlayoutMode() const override;
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// Writes the current network statistics to |stats|. The statistics are reset
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// after the call.
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int NetworkStatistics(NetEqNetworkStatistics* stats) override;
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// Writes the current RTCP statistics to |stats|. The statistics are reset
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// and a new report period is started with the call.
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void GetRtcpStatistics(RtcpStatistics* stats) override;
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NetEqLifetimeStatistics GetLifetimeStatistics() const override;
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// Same as RtcpStatistics(), but does not reset anything.
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void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
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// Enables post-decode VAD. When enabled, GetAudio() will return
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// kOutputVADPassive when the signal contains no speech.
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void EnableVad() override;
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// Disables post-decode VAD.
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void DisableVad() override;
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rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
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int last_output_sample_rate_hz() const override;
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rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
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rtc::Optional<SdpAudioFormat> GetDecoderFormat(
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int payload_type) const override;
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int SetTargetNumberOfChannels() override;
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int SetTargetSampleRate() override;
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// Flushes both the packet buffer and the sync buffer.
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void FlushBuffers() override;
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void PacketBufferStatistics(int* current_num_packets,
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int* max_num_packets) const override;
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void EnableNack(size_t max_nack_list_size) override;
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void DisableNack() override;
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std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
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std::vector<uint32_t> LastDecodedTimestamps() const override;
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int SyncBufferSizeMs() const override;
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// This accessor method is only intended for testing purposes.
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const SyncBuffer* sync_buffer_for_test() const;
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Operations last_operation_for_test() const;
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protected:
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static const int kOutputSizeMs = 10;
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static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
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// TODO(hlundin): Provide a better value for kSyncBufferSize.
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// Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
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// calculating correlations of current frame against history.
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static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
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// Inserts a new packet into NetEq. This is used by the InsertPacket method
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// above. Returns 0 on success, otherwise an error code.
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// TODO(hlundin): Merge this with InsertPacket above?
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int InsertPacketInternal(const RTPHeader& rtp_header,
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rtc::ArrayView<const uint8_t> payload,
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uint32_t receive_timestamp)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Delivers 10 ms of audio data. The data is written to |audio_frame|.
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// Returns 0 on success, otherwise an error code.
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int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Provides a decision to the GetAudioInternal method. The decision what to
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// do is written to |operation|. Packets to decode are written to
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// |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
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// DTMF should be played, |play_dtmf| is set to true by the method.
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// Returns 0 on success, otherwise an error code.
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int GetDecision(Operations* operation,
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PacketList* packet_list,
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DtmfEvent* dtmf_event,
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bool* play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Decodes the speech packets in |packet_list|, and writes the results to
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// |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
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// elements. The length of the decoded data is written to |decoded_length|.
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// The speech type -- speech or (codec-internal) comfort noise -- is written
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// to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
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// comfort noise, those are not decoded.
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int Decode(PacketList* packet_list,
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Operations* operation,
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int* decoded_length,
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AudioDecoder::SpeechType* speech_type)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method to Decode(). Performs codec internal CNG.
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int DecodeCng(AudioDecoder* decoder,
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int* decoded_length,
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AudioDecoder::SpeechType* speech_type)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method to Decode(). Performs the actual decoding.
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int DecodeLoop(PacketList* packet_list,
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const Operations& operation,
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AudioDecoder* decoder,
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int* decoded_length,
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AudioDecoder::SpeechType* speech_type)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the Normal class to perform the normal operation.
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void DoNormal(const int16_t* decoded_buffer,
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size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the Merge class to perform the merge operation.
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void DoMerge(int16_t* decoded_buffer,
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size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the Expand class to perform the expand operation.
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int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the Accelerate class to perform the accelerate
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// operation.
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int DoAccelerate(int16_t* decoded_buffer,
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size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf,
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bool fast_accelerate)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the PreemptiveExpand class to perform the
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// preemtive expand operation.
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int DoPreemptiveExpand(int16_t* decoded_buffer,
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size_t decoded_length,
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AudioDecoder::SpeechType speech_type,
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bool play_dtmf)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
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// noise. |packet_list| can either contain one SID frame to update the
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// noise parameters, or no payload at all, in which case the previously
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// received parameters are used.
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int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Calls the audio decoder to generate codec-internal comfort noise when
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// no packet was received.
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void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Calls the DtmfToneGenerator class to generate DTMF tones.
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int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Produces packet-loss concealment using alternative methods. If the codec
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// has an internal PLC, it is called to generate samples. Otherwise, the
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// method performs zero-stuffing.
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void DoAlternativePlc(bool increase_timestamp)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Overdub DTMF on top of |output|.
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int DtmfOverdub(const DtmfEvent& dtmf_event,
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size_t num_channels,
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int16_t* output) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Extracts packets from |packet_buffer_| to produce at least
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// |required_samples| samples. The packets are inserted into |packet_list|.
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// Returns the number of samples that the packets in the list will produce, or
|
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// -1 in case of an error.
|
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int ExtractPackets(size_t required_samples, PacketList* packet_list)
|
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Resets various variables and objects to new values based on the sample rate
|
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// |fs_hz| and |channels| number audio channels.
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void SetSampleRateAndChannels(int fs_hz, size_t channels)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Returns the output type for the audio produced by the latest call to
|
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// GetAudio().
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OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
|
||||
|
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// Updates Expand and Merge.
|
||||
virtual void UpdatePlcComponents(int fs_hz, size_t channels)
|
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RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
|
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// Creates DecisionLogic object with the mode given by |playout_mode_|.
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virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
|
||||
|
||||
rtc::CriticalSection crit_sect_;
|
||||
const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<DecoderDatabase> decoder_database_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<TimestampScaler> timestamp_scaler_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<ExpandFactory> expand_factory_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<AccelerateFactory> accelerate_factory_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
|
||||
std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<AudioMultiVector> algorithm_buffer_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<PreemptiveExpand> preemptive_expand_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
|
||||
Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
|
||||
StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
|
||||
int fs_hz_ RTC_GUARDED_BY(crit_sect_);
|
||||
int fs_mult_ RTC_GUARDED_BY(crit_sect_);
|
||||
int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
|
||||
size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
|
||||
size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
|
||||
Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
|
||||
Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<int16_t[]> mute_factor_array_ RTC_GUARDED_BY(crit_sect_);
|
||||
size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
|
||||
uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
|
||||
bool new_codec_ RTC_GUARDED_BY(crit_sect_);
|
||||
uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
|
||||
bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
|
||||
rtc::Optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
|
||||
rtc::Optional<uint8_t> current_cng_rtp_payload_type_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
|
||||
bool first_packet_ RTC_GUARDED_BY(crit_sect_);
|
||||
const BackgroundNoiseMode background_noise_mode_ RTC_GUARDED_BY(crit_sect_);
|
||||
NetEqPlayoutMode playout_mode_ RTC_GUARDED_BY(crit_sect_);
|
||||
bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
|
||||
std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
|
||||
bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
|
||||
const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
|
||||
AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
|
||||
AudioFrame::kVadPassive;
|
||||
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
|
||||
RTC_GUARDED_BY(crit_sect_);
|
||||
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
|
||||
|
||||
private:
|
||||
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
|
||||
Reference in New Issue
Block a user