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
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
new file mode 100644
index 0000000..e92babd
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -0,0 +1,406 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * 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_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq/defines.h"
+#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
+#include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/rtcp.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Accelerate;
+class BackgroundNoise;
+class BufferLevelFilter;
+class ComfortNoise;
+class CriticalSectionWrapper;
+class DecisionLogic;
+class DecoderDatabase;
+class DelayManager;
+class DelayPeakDetector;
+class DtmfBuffer;
+class DtmfToneGenerator;
+class Expand;
+class Merge;
+class Normal;
+class PacketBuffer;
+class PayloadSplitter;
+class PostDecodeVad;
+class PreemptiveExpand;
+class RandomVector;
+class SyncBuffer;
+class TimestampScaler;
+struct AccelerateFactory;
+struct DtmfEvent;
+struct ExpandFactory;
+struct PreemptiveExpandFactory;
+
+class NetEqImpl : public webrtc::NetEq {
+ public:
+ // Creates a new NetEqImpl object. The object will assume ownership of all
+ // injected dependencies, and will delete them when done.
+ NetEqImpl(int fs,
+ BufferLevelFilter* buffer_level_filter,
+ DecoderDatabase* decoder_database,
+ DelayManager* delay_manager,
+ DelayPeakDetector* delay_peak_detector,
+ DtmfBuffer* dtmf_buffer,
+ DtmfToneGenerator* dtmf_tone_generator,
+ PacketBuffer* packet_buffer,
+ PayloadSplitter* payload_splitter,
+ TimestampScaler* timestamp_scaler,
+ AccelerateFactory* accelerate_factory,
+ ExpandFactory* expand_factory,
+ PreemptiveExpandFactory* preemptive_expand_factory,
+ bool create_components = true);
+
+ virtual ~NetEqImpl();
+
+ // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
+ // of the time when the packet was received, and should be measured with
+ // the same tick rate as the RTP timestamp of the current payload.
+ // Returns 0 on success, -1 on failure.
+ virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
+ const uint8_t* payload,
+ int length_bytes,
+ uint32_t receive_timestamp);
+
+ // Inserts a sync-packet into packet queue. Sync-packets are decoded to
+ // silence and are intended to keep AV-sync intact in an event of long packet
+ // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
+ // might insert sync-packet when they observe that buffer level of NetEq is
+ // decreasing below a certain threshold, defined by the application.
+ // Sync-packets should have the same payload type as the last audio payload
+ // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
+ // can be implied by inserting a sync-packet.
+ // Returns kOk on success, kFail on failure.
+ virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
+ uint32_t receive_timestamp);
+
+ // Instructs NetEq to deliver 10 ms of audio data. The data is written to
+ // |output_audio|, which can hold (at least) |max_length| elements.
+ // The number of channels that were written to the output is provided in
+ // the output variable |num_channels|, and each channel contains
+ // |samples_per_channel| elements. If more than one channel is written,
+ // the samples are interleaved.
+ // The speech type is written to |type|, if |type| is not NULL.
+ // Returns kOK on success, or kFail in case of an error.
+ virtual int GetAudio(size_t max_length, int16_t* output_audio,
+ int* samples_per_channel, int* num_channels,
+ NetEqOutputType* type);
+
+ // Associates |rtp_payload_type| with |codec| and stores the information in
+ // the codec database. Returns kOK on success, kFail on failure.
+ virtual int RegisterPayloadType(enum NetEqDecoder codec,
+ uint8_t rtp_payload_type);
+
+ // Provides an externally created decoder object |decoder| to insert in the
+ // decoder database. The decoder implements a decoder of type |codec| and
+ // associates it with |rtp_payload_type|. Returns kOK on success, kFail on
+ // failure.
+ virtual int RegisterExternalDecoder(AudioDecoder* decoder,
+ enum NetEqDecoder codec,
+ uint8_t rtp_payload_type);
+
+ // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
+ // -1 on failure.
+ virtual int RemovePayloadType(uint8_t rtp_payload_type);
+
+ virtual bool SetMinimumDelay(int delay_ms);
+
+ virtual bool SetMaximumDelay(int delay_ms);
+
+ virtual int LeastRequiredDelayMs() const;
+
+ virtual int SetTargetDelay() { return kNotImplemented; }
+
+ virtual int TargetDelay() { return kNotImplemented; }
+
+ virtual int CurrentDelay() { return kNotImplemented; }
+
+ // Sets the playout mode to |mode|.
+ virtual void SetPlayoutMode(NetEqPlayoutMode mode);
+
+ // Returns the current playout mode.
+ virtual NetEqPlayoutMode PlayoutMode() const;
+
+ // Writes the current network statistics to |stats|. The statistics are reset
+ // after the call.
+ virtual int NetworkStatistics(NetEqNetworkStatistics* stats);
+
+ // Writes the last packet waiting times (in ms) to |waiting_times|. The number
+ // of values written is no more than 100, but may be smaller if the interface
+ // is polled again before 100 packets has arrived.
+ virtual void WaitingTimes(std::vector<int>* waiting_times);
+
+ // Writes the current RTCP statistics to |stats|. The statistics are reset
+ // and a new report period is started with the call.
+ virtual void GetRtcpStatistics(RtcpStatistics* stats);
+
+ // Same as RtcpStatistics(), but does not reset anything.
+ virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats);
+
+ // Enables post-decode VAD. When enabled, GetAudio() will return
+ // kOutputVADPassive when the signal contains no speech.
+ virtual void EnableVad();
+
+ // Disables post-decode VAD.
+ virtual void DisableVad();
+
+ virtual bool GetPlayoutTimestamp(uint32_t* timestamp);
+
+ virtual int SetTargetNumberOfChannels() { return kNotImplemented; }
+
+ virtual int SetTargetSampleRate() { return kNotImplemented; }
+
+ // Returns the error code for the last occurred error. If no error has
+ // occurred, 0 is returned.
+ virtual int LastError();
+
+ // Returns the error code last returned by a decoder (audio or comfort noise).
+ // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
+ // this method to get the decoder's error code.
+ virtual int LastDecoderError();
+
+ // Flushes both the packet buffer and the sync buffer.
+ virtual void FlushBuffers();
+
+ virtual void PacketBufferStatistics(int* current_num_packets,
+ int* max_num_packets) const;
+
+ // Get sequence number and timestamp of the latest RTP.
+ // This method is to facilitate NACK.
+ virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const;
+
+ // Sets background noise mode.
+ virtual void SetBackgroundNoiseMode(NetEqBackgroundNoiseMode mode);
+
+ // Gets background noise mode.
+ virtual NetEqBackgroundNoiseMode BackgroundNoiseMode() const;
+
+ // This accessor method is only intended for testing purposes.
+ virtual const SyncBuffer* sync_buffer_for_test() const;
+
+ protected:
+ static const int kOutputSizeMs = 10;
+ static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
+ // TODO(hlundin): Provide a better value for kSyncBufferSize.
+ static const int kSyncBufferSize = 2 * kMaxFrameSize;
+
+ // Inserts a new packet into NetEq. This is used by the InsertPacket method
+ // above. Returns 0 on success, otherwise an error code.
+ // TODO(hlundin): Merge this with InsertPacket above?
+ int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
+ const uint8_t* payload,
+ int length_bytes,
+ uint32_t receive_timestamp,
+ bool is_sync_packet)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Delivers 10 ms of audio data. The data is written to |output|, which can
+ // hold (at least) |max_length| elements. The number of channels that were
+ // written to the output is provided in the output variable |num_channels|,
+ // and each channel contains |samples_per_channel| elements. If more than one
+ // channel is written, the samples are interleaved.
+ // Returns 0 on success, otherwise an error code.
+ int GetAudioInternal(size_t max_length,
+ int16_t* output,
+ int* samples_per_channel,
+ int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Provides a decision to the GetAudioInternal method. The decision what to
+ // do is written to |operation|. Packets to decode are written to
+ // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
+ // DTMF should be played, |play_dtmf| is set to true by the method.
+ // Returns 0 on success, otherwise an error code.
+ int GetDecision(Operations* operation,
+ PacketList* packet_list,
+ DtmfEvent* dtmf_event,
+ bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Decodes the speech packets in |packet_list|, and writes the results to
+ // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
+ // elements. The length of the decoded data is written to |decoded_length|.
+ // The speech type -- speech or (codec-internal) comfort noise -- is written
+ // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
+ // comfort noise, those are not decoded.
+ int Decode(PacketList* packet_list,
+ Operations* operation,
+ int* decoded_length,
+ AudioDecoder::SpeechType* speech_type)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method to Decode(). Performs the actual decoding.
+ int DecodeLoop(PacketList* packet_list,
+ Operations* operation,
+ AudioDecoder* decoder,
+ int* decoded_length,
+ AudioDecoder::SpeechType* speech_type)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the Normal class to perform the normal operation.
+ void DoNormal(const int16_t* decoded_buffer,
+ size_t decoded_length,
+ AudioDecoder::SpeechType speech_type,
+ bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the Merge class to perform the merge operation.
+ void DoMerge(int16_t* decoded_buffer,
+ size_t decoded_length,
+ AudioDecoder::SpeechType speech_type,
+ bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the Expand class to perform the expand operation.
+ int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the Accelerate class to perform the accelerate
+ // operation.
+ int DoAccelerate(int16_t* decoded_buffer,
+ size_t decoded_length,
+ AudioDecoder::SpeechType speech_type,
+ bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the PreemptiveExpand class to perform the
+ // preemtive expand operation.
+ int DoPreemptiveExpand(int16_t* decoded_buffer,
+ size_t decoded_length,
+ AudioDecoder::SpeechType speech_type,
+ bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
+ // noise. |packet_list| can either contain one SID frame to update the
+ // noise parameters, or no payload at all, in which case the previously
+ // received parameters are used.
+ int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Calls the audio decoder to generate codec-internal comfort noise when
+ // no packet was received.
+ void DoCodecInternalCng() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Calls the DtmfToneGenerator class to generate DTMF tones.
+ int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Produces packet-loss concealment using alternative methods. If the codec
+ // has an internal PLC, it is called to generate samples. Otherwise, the
+ // method performs zero-stuffing.
+ void DoAlternativePlc(bool increase_timestamp)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Overdub DTMF on top of |output|.
+ int DtmfOverdub(const DtmfEvent& dtmf_event,
+ size_t num_channels,
+ int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Extracts packets from |packet_buffer_| to produce at least
+ // |required_samples| samples. The packets are inserted into |packet_list|.
+ // Returns the number of samples that the packets in the list will produce, or
+ // -1 in case of an error.
+ int ExtractPackets(int required_samples, PacketList* packet_list)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Resets various variables and objects to new values based on the sample rate
+ // |fs_hz| and |channels| number audio channels.
+ void SetSampleRateAndChannels(int fs_hz, size_t channels)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Returns the output type for the audio produced by the latest call to
+ // GetAudio().
+ NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Updates Expand and Merge.
+ virtual void UpdatePlcComponents(int fs_hz, size_t channels)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ // Creates DecisionLogic object for the given mode.
+ virtual void CreateDecisionLogic(NetEqPlayoutMode mode)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ const scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ const scoped_ptr<BufferLevelFilter> buffer_level_filter_
+ GUARDED_BY(crit_sect_);
+ const scoped_ptr<DecoderDatabase> decoder_database_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<DelayPeakDetector> delay_peak_detector_
+ GUARDED_BY(crit_sect_);
+ const scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_
+ GUARDED_BY(crit_sect_);
+ const scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<PayloadSplitter> payload_splitter_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<TimestampScaler> timestamp_scaler_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
+ const scoped_ptr<AccelerateFactory> accelerate_factory_
+ GUARDED_BY(crit_sect_);
+ const scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
+ GUARDED_BY(crit_sect_);
+
+ scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
+ scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
+ scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
+ scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
+ scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
+ scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
+ 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_);
+ 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_