Adding thread annotations to parts of Audio Coding Module
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.
BUG=3401
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index b8bb12c..7f6c840 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -116,7 +116,7 @@
AudioCodingModuleImpl::AudioCodingModuleImpl(
const AudioCodingModule::Config& config)
- : packetization_callback_(NULL),
+ : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
id_(config.id),
expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
@@ -133,8 +133,6 @@
current_send_codec_idx_(-1),
send_codec_registered_(false),
receiver_(config),
- acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- vad_callback_(NULL),
is_first_red_(true),
red_enabled_(false),
last_red_timestamp_(0),
@@ -142,10 +140,12 @@
previous_pltype_(255),
aux_rtp_header_(NULL),
receiver_initialized_(false),
- callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
secondary_send_codec_inst_(),
codec_timestamp_(expected_codec_ts_),
- first_10ms_data_(false) {
+ first_10ms_data_(false),
+ callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+ packetization_callback_(NULL),
+ vad_callback_(NULL) {
// Nullify send codec memory, set payload type and set codec name to
// invalid values.
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 247aee7..e54202b 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -19,11 +19,11 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
namespace webrtc {
class CriticalSectionWrapper;
-class RWLockWrapper;
namespace acm2 {
@@ -246,13 +246,14 @@
ACMGenericCodec* CreateCodec(const CodecInst& codec);
- int InitializeReceiverSafe();
+ int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
bool HaveValidEncoder(const char* caller_name) const;
// Set VAD/DTX status. This function does not acquire a lock, and it is
// created to be called only from inside a critical section.
- int SetVADSafe(bool enable_dtx, bool enable_vad, ACMVADMode mode);
+ int SetVADSafe(bool enable_dtx, bool enable_vad, ACMVADMode mode)
+ EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
// Process buffered audio when dual-streaming is not enabled (When RED is
// enabled still this function is used.)
@@ -274,18 +275,22 @@
// -1: if encountering an error.
// 0: otherwise.
int PreprocessToAddData(const AudioFrame& in_frame,
- const AudioFrame** ptr_out);
+ const AudioFrame** ptr_out)
+ EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
// Change required states after starting to receive the codec corresponding
// to |index|.
int UpdateUponReceivingCodec(int index);
- int EncodeFragmentation(int fragmentation_index, int payload_type,
+ int EncodeFragmentation(int fragmentation_index,
+ int payload_type,
uint32_t current_timestamp,
ACMGenericCodec* encoder,
- uint8_t* stream);
+ uint8_t* stream)
+ EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
- void ResetFragmentation(int vector_size);
+ void ResetFragmentation(int vector_size)
+ EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
// Get a pointer to AudioDecoder of the given codec. For some codecs, e.g.
// iSAC, encoding and decoding have to be performed on a shared
@@ -300,53 +305,50 @@
int GetAudioDecoder(const CodecInst& codec, int codec_id,
int mirror_id, AudioDecoder** decoder);
- AudioPacketizationCallback* packetization_callback_;
+ CriticalSectionWrapper* acm_crit_sect_;
+ int id_; // TODO(henrik.lundin) Make const.
+ uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
+ uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
+ CodecInst send_codec_inst_ GUARDED_BY(acm_crit_sect_);
- int id_;
- uint32_t expected_codec_ts_;
- uint32_t expected_in_ts_;
- CodecInst send_codec_inst_;
+ uint8_t cng_nb_pltype_ GUARDED_BY(acm_crit_sect_);
+ uint8_t cng_wb_pltype_ GUARDED_BY(acm_crit_sect_);
+ uint8_t cng_swb_pltype_ GUARDED_BY(acm_crit_sect_);
+ uint8_t cng_fb_pltype_ GUARDED_BY(acm_crit_sect_);
- uint8_t cng_nb_pltype_;
- uint8_t cng_wb_pltype_;
- uint8_t cng_swb_pltype_;
- uint8_t cng_fb_pltype_;
-
- uint8_t red_pltype_;
- bool vad_enabled_;
- bool dtx_enabled_;
- ACMVADMode vad_mode_;
+ uint8_t red_pltype_ GUARDED_BY(acm_crit_sect_);
+ bool vad_enabled_ GUARDED_BY(acm_crit_sect_);
+ bool dtx_enabled_ GUARDED_BY(acm_crit_sect_);
+ ACMVADMode vad_mode_ GUARDED_BY(acm_crit_sect_);
ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs];
int mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs];
- bool stereo_send_;
+ bool stereo_send_ GUARDED_BY(acm_crit_sect_);
int current_send_codec_idx_;
bool send_codec_registered_;
- ACMResampler resampler_;
+ ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
AcmReceiver receiver_;
- CriticalSectionWrapper* acm_crit_sect_;
- ACMVADCallback* vad_callback_;
// RED.
- bool is_first_red_;
- bool red_enabled_;
+ bool is_first_red_ GUARDED_BY(acm_crit_sect_);
+ bool red_enabled_ GUARDED_BY(acm_crit_sect_);
// TODO(turajs): |red_buffer_| is allocated in constructor, why having them
// as pointers and not an array. If concerned about the memory, then make a
// set-up function to allocate them only when they are going to be used, i.e.
// RED or Dual-streaming is enabled.
- uint8_t* red_buffer_;
+ uint8_t* red_buffer_ GUARDED_BY(acm_crit_sect_);
// TODO(turajs): we actually don't need |fragmentation_| as a member variable.
// It is sufficient to keep the length & payload type of previous payload in
// member variables.
- RTPFragmentationHeader fragmentation_;
- uint32_t last_red_timestamp_;
+ RTPFragmentationHeader fragmentation_ GUARDED_BY(acm_crit_sect_);
+ uint32_t last_red_timestamp_ GUARDED_BY(acm_crit_sect_);
// Codec internal FEC
bool codec_fec_enabled_;
// This is to keep track of CN instances where we can send DTMFs.
- uint8_t previous_pltype_;
+ uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
// Used when payloads are pushed into ACM without any RTP info
// One example is when pre-encoded bit-stream is pushed from
@@ -356,15 +358,18 @@
// be used in other methods, locks need to be taken.
WebRtcRTPHeader* aux_rtp_header_;
- bool receiver_initialized_;
+ bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
+
+ AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
+ CodecInst secondary_send_codec_inst_ GUARDED_BY(acm_crit_sect_);
+ scoped_ptr<ACMGenericCodec> secondary_encoder_ GUARDED_BY(acm_crit_sect_);
+ uint32_t codec_timestamp_ GUARDED_BY(acm_crit_sect_);
+ bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
CriticalSectionWrapper* callback_crit_sect_;
-
- AudioFrame preprocess_frame_;
- CodecInst secondary_send_codec_inst_;
- scoped_ptr<ACMGenericCodec> secondary_encoder_;
- uint32_t codec_timestamp_;
- bool first_10ms_data_;
+ AudioPacketizationCallback* packetization_callback_
+ GUARDED_BY(callback_crit_sect_);
+ ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
};
} // namespace acm2
