Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
406
webrtc/modules/audio_coding/neteq/neteq_impl.h
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406
webrtc/modules/audio_coding/neteq/neteq_impl.h
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@ -0,0 +1,406 @@
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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 <vector>
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#include "webrtc/base/constructormagic.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/interface/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/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/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 CriticalSectionWrapper;
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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 Normal;
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class PacketBuffer;
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class PayloadSplitter;
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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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// Creates a new NetEqImpl object. The object will assume ownership of all
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// injected dependencies, and will delete them when done.
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NetEqImpl(int fs,
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BufferLevelFilter* buffer_level_filter,
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DecoderDatabase* decoder_database,
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DelayManager* delay_manager,
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DelayPeakDetector* delay_peak_detector,
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DtmfBuffer* dtmf_buffer,
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DtmfToneGenerator* dtmf_tone_generator,
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PacketBuffer* packet_buffer,
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PayloadSplitter* payload_splitter,
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TimestampScaler* timestamp_scaler,
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AccelerateFactory* accelerate_factory,
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ExpandFactory* expand_factory,
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PreemptiveExpandFactory* preemptive_expand_factory,
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bool create_components = true);
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virtual ~NetEqImpl();
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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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virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
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const uint8_t* payload,
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int length_bytes,
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uint32_t receive_timestamp);
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// Inserts a sync-packet into packet queue. Sync-packets are decoded to
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// silence and are intended to keep AV-sync intact in an event of long packet
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// losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
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// might insert sync-packet when they observe that buffer level of NetEq is
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// decreasing below a certain threshold, defined by the application.
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// Sync-packets should have the same payload type as the last audio payload
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// type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
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// can be implied by inserting a sync-packet.
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// Returns kOk on success, kFail on failure.
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virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
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uint32_t receive_timestamp);
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// Instructs NetEq to deliver 10 ms of audio data. The data is written to
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// |output_audio|, which can hold (at least) |max_length| elements.
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// The number of channels that were written to the output is provided in
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// the output variable |num_channels|, and each channel contains
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// |samples_per_channel| elements. If more than one channel is written,
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// the samples are interleaved.
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// The speech type is written to |type|, if |type| is not NULL.
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// Returns kOK on success, or kFail in case of an error.
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virtual int GetAudio(size_t max_length, int16_t* output_audio,
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int* samples_per_channel, int* num_channels,
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NetEqOutputType* type);
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// Associates |rtp_payload_type| with |codec| and stores the information in
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// the codec database. Returns kOK on success, kFail on failure.
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virtual int RegisterPayloadType(enum NetEqDecoder codec,
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uint8_t rtp_payload_type);
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// Provides an externally created decoder object |decoder| to insert in the
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// decoder database. The decoder implements a decoder of type |codec| and
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// associates it with |rtp_payload_type|. Returns kOK on success, kFail on
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// failure.
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virtual int RegisterExternalDecoder(AudioDecoder* decoder,
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enum NetEqDecoder codec,
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uint8_t rtp_payload_type);
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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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virtual int RemovePayloadType(uint8_t rtp_payload_type);
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virtual bool SetMinimumDelay(int delay_ms);
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virtual bool SetMaximumDelay(int delay_ms);
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virtual int LeastRequiredDelayMs() const;
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virtual int SetTargetDelay() { return kNotImplemented; }
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virtual int TargetDelay() { return kNotImplemented; }
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virtual int CurrentDelay() { return kNotImplemented; }
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// Sets the playout mode to |mode|.
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virtual void SetPlayoutMode(NetEqPlayoutMode mode);
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// Returns the current playout mode.
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virtual NetEqPlayoutMode PlayoutMode() const;
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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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virtual int NetworkStatistics(NetEqNetworkStatistics* stats);
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// Writes the last packet waiting times (in ms) to |waiting_times|. The number
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// of values written is no more than 100, but may be smaller if the interface
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// is polled again before 100 packets has arrived.
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virtual void WaitingTimes(std::vector<int>* waiting_times);
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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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virtual void GetRtcpStatistics(RtcpStatistics* stats);
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// Same as RtcpStatistics(), but does not reset anything.
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virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats);
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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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virtual void EnableVad();
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// Disables post-decode VAD.
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virtual void DisableVad();
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virtual bool GetPlayoutTimestamp(uint32_t* timestamp);
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virtual int SetTargetNumberOfChannels() { return kNotImplemented; }
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virtual int SetTargetSampleRate() { return kNotImplemented; }
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// Returns the error code for the last occurred error. If no error has
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// occurred, 0 is returned.
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virtual int LastError();
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// Returns the error code last returned by a decoder (audio or comfort noise).
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// When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
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// this method to get the decoder's error code.
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virtual int LastDecoderError();
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// Flushes both the packet buffer and the sync buffer.
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virtual void FlushBuffers();
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virtual void PacketBufferStatistics(int* current_num_packets,
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int* max_num_packets) const;
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// Get sequence number and timestamp of the latest RTP.
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// This method is to facilitate NACK.
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virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const;
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// Sets background noise mode.
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virtual void SetBackgroundNoiseMode(NetEqBackgroundNoiseMode mode);
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// Gets background noise mode.
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virtual NetEqBackgroundNoiseMode BackgroundNoiseMode() const;
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// This accessor method is only intended for testing purposes.
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virtual const SyncBuffer* sync_buffer_for_test() const;
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protected:
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static const int kOutputSizeMs = 10;
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static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
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// TODO(hlundin): Provide a better value for kSyncBufferSize.
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static const int kSyncBufferSize = 2 * kMaxFrameSize;
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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 WebRtcRTPHeader& rtp_header,
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const uint8_t* payload,
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int length_bytes,
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uint32_t receive_timestamp,
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bool is_sync_packet)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Delivers 10 ms of audio data. The data is written to |output|, which can
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// hold (at least) |max_length| elements. The number of channels that were
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// written to the output is provided in the output variable |num_channels|,
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// and each channel contains |samples_per_channel| elements. If more than one
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// channel is written, the samples are interleaved.
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// Returns 0 on success, otherwise an error code.
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int GetAudioInternal(size_t max_length,
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int16_t* output,
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int* samples_per_channel,
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int* num_channels) 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) 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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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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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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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) 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) 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) 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) 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) 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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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() 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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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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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 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(int required_samples, PacketList* packet_list)
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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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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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NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Updates Expand and Merge.
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virtual void UpdatePlcComponents(int fs_hz, size_t channels)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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// Creates DecisionLogic object for the given mode.
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virtual void CreateDecisionLogic(NetEqPlayoutMode mode)
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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const scoped_ptr<CriticalSectionWrapper> crit_sect_;
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const scoped_ptr<BufferLevelFilter> buffer_level_filter_
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GUARDED_BY(crit_sect_);
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const scoped_ptr<DecoderDatabase> decoder_database_ GUARDED_BY(crit_sect_);
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const scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
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const scoped_ptr<DelayPeakDetector> delay_peak_detector_
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GUARDED_BY(crit_sect_);
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const scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
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const scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_
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GUARDED_BY(crit_sect_);
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const scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
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const scoped_ptr<PayloadSplitter> payload_splitter_ GUARDED_BY(crit_sect_);
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const scoped_ptr<TimestampScaler> timestamp_scaler_ GUARDED_BY(crit_sect_);
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const scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
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const scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
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const scoped_ptr<AccelerateFactory> accelerate_factory_
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GUARDED_BY(crit_sect_);
|
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const scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
|
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GUARDED_BY(crit_sect_);
|
||||
|
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scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
|
||||
scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
|
||||
scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
|
||||
RandomVector random_vector_ GUARDED_BY(crit_sect_);
|
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scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
|
||||
Rtcp rtcp_ GUARDED_BY(crit_sect_);
|
||||
StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
|
||||
int fs_hz_ GUARDED_BY(crit_sect_);
|
||||
int fs_mult_ GUARDED_BY(crit_sect_);
|
||||
int output_size_samples_ GUARDED_BY(crit_sect_);
|
||||
int decoder_frame_length_ GUARDED_BY(crit_sect_);
|
||||
Modes last_mode_ GUARDED_BY(crit_sect_);
|
||||
scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
|
||||
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
|
||||
scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
|
||||
uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
|
||||
bool new_codec_ GUARDED_BY(crit_sect_);
|
||||
uint32_t timestamp_ GUARDED_BY(crit_sect_);
|
||||
bool reset_decoder_ GUARDED_BY(crit_sect_);
|
||||
uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
|
||||
uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
|
||||
uint32_t ssrc_ GUARDED_BY(crit_sect_);
|
||||
bool first_packet_ GUARDED_BY(crit_sect_);
|
||||
int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
|
||||
int decoder_error_code_ GUARDED_BY(crit_sect_);
|
||||
|
||||
// These values are used by NACK module to estimate time-to-play of
|
||||
// a missing packet. Occasionally, NetEq might decide to decode more
|
||||
// than one packet. Therefore, these values store sequence number and
|
||||
// timestamp of the first packet pulled from the packet buffer. In
|
||||
// such cases, these values do not exactly represent the sequence number
|
||||
// or timestamp associated with a 10ms audio pulled from NetEq. NACK
|
||||
// module is designed to compensate for this.
|
||||
int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
|
||||
uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
|
||||
|
||||
private:
|
||||
DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
|
||||
Reference in New Issue
Block a user