Adding thread annotations to NetEq4
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5716 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
index c17ff1e..511452e 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
@@ -22,6 +22,7 @@
#include "webrtc/modules/audio_coding/neteq4/statistics_calculator.h"
#include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -212,8 +213,8 @@
const uint8_t* payload,
int length_bytes,
uint32_t receive_timestamp,
- bool is_sync_packet);
-
+ 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
@@ -221,9 +222,10 @@
// 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);
-
+ 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
@@ -233,7 +235,7 @@
int GetDecision(Operations* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
- bool* play_dtmf);
+ 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|
@@ -241,116 +243,137 @@
// 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);
+ 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);
+ 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);
+ 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);
+ 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);
+ 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);
+ 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);
+ 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);
+ 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();
+ void DoCodecInternalCng() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Calls the DtmfToneGenerator class to generate DTMF tones.
- int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf);
+ 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);
+ 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;
+ 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);
+ 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);
+ 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();
+ NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
- scoped_ptr<BackgroundNoise> background_noise_;
- scoped_ptr<BufferLevelFilter> buffer_level_filter_;
- scoped_ptr<DecoderDatabase> decoder_database_;
- scoped_ptr<DelayManager> delay_manager_;
- scoped_ptr<DelayPeakDetector> delay_peak_detector_;
- scoped_ptr<DtmfBuffer> dtmf_buffer_;
- scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_;
- scoped_ptr<PacketBuffer> packet_buffer_;
- scoped_ptr<PayloadSplitter> payload_splitter_;
- scoped_ptr<TimestampScaler> timestamp_scaler_;
- scoped_ptr<DecisionLogic> decision_logic_;
- scoped_ptr<PostDecodeVad> vad_;
- scoped_ptr<AudioMultiVector> algorithm_buffer_;
- scoped_ptr<SyncBuffer> sync_buffer_;
- scoped_ptr<Expand> expand_;
- scoped_ptr<ExpandFactory> expand_factory_;
- scoped_ptr<Normal> normal_;
- scoped_ptr<Merge> merge_;
- scoped_ptr<Accelerate> accelerate_;
- scoped_ptr<AccelerateFactory> accelerate_factory_;
- scoped_ptr<PreemptiveExpand> preemptive_expand_;
- scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_;
- RandomVector random_vector_;
- scoped_ptr<ComfortNoise> comfort_noise_;
- Rtcp rtcp_;
- StatisticsCalculator stats_;
- int fs_hz_;
- int fs_mult_;
- int output_size_samples_;
- int decoder_frame_length_;
- Modes last_mode_;
- scoped_array<int16_t> mute_factor_array_;
- size_t decoded_buffer_length_;
- scoped_array<int16_t> decoded_buffer_;
- uint32_t playout_timestamp_;
- bool new_codec_;
- uint32_t timestamp_;
- bool reset_decoder_;
- uint8_t current_rtp_payload_type_;
- uint8_t current_cng_rtp_payload_type_;
- uint32_t ssrc_;
- bool first_packet_;
- int error_code_; // Store last error code.
- int decoder_error_code_;
- scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ const scoped_ptr<BufferLevelFilter> buffer_level_filter_;
+ const scoped_ptr<DecoderDatabase> decoder_database_;
+ const scoped_ptr<DelayManager> delay_manager_;
+ const scoped_ptr<DelayPeakDetector> delay_peak_detector_;
+ const scoped_ptr<DtmfBuffer> dtmf_buffer_;
+ const scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_;
+ const scoped_ptr<PacketBuffer> packet_buffer_;
+ const scoped_ptr<PayloadSplitter> payload_splitter_;
+ const scoped_ptr<TimestampScaler> timestamp_scaler_;
+ const scoped_ptr<PostDecodeVad> vad_;
+ const scoped_ptr<ExpandFactory> expand_factory_;
+ const scoped_ptr<AccelerateFactory> accelerate_factory_;
+ const scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_;
+
+ 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_array<int16_t> mute_factor_array_ GUARDED_BY(crit_sect_);
+ size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
+ scoped_array<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_);
+ const scoped_ptr<CriticalSectionWrapper> 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
@@ -359,8 +382,8 @@
// 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_;
- uint32_t decoded_packet_timestamp_;
+ int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
+ uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
};