Delete AcmReceiver::SetInitialDelay
This method is no longer called. With that gone, a number of other
methods and member variables are obsoleted, and removed.
Methods deleted:
AcmReceiver::InsertStreamOfSyncPackets
AcmReceiver::GetNumSyncPacketToInsert()
AcmReceiver::GetSilence, never called
Member variables deleted:
missing_packets_sync_stream_
late_packets_sync_stream_
av_sync_
initial_delay_manager_
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1419573013
Cr-Commit-Position: refs/heads/master@{#10484}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 209d816..bc9826a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -21,7 +21,6 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@@ -130,11 +129,7 @@
neteq_(NetEq::Create(config.neteq_config)),
vad_enabled_(config.neteq_config.enable_post_decode_vad),
clock_(config.clock),
- resampled_last_output_frame_(true),
- av_sync_(false),
- initial_delay_manager_(),
- missing_packets_sync_stream_(),
- late_packets_sync_stream_() {
+ resampled_last_output_frame_(true) {
assert(clock_);
memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
@@ -151,42 +146,6 @@
return -1;
}
-int AcmReceiver::SetInitialDelay(int delay_ms) {
- if (delay_ms < 0 || delay_ms > 10000) {
- return -1;
- }
- CriticalSectionScoped lock(crit_sect_.get());
-
- if (delay_ms == 0) {
- av_sync_ = false;
- initial_delay_manager_.reset();
- missing_packets_sync_stream_.reset();
- late_packets_sync_stream_.reset();
- neteq_->SetMinimumDelay(0);
- return 0;
- }
-
- if (av_sync_ && initial_delay_manager_->PacketBuffered()) {
- // Too late for this API. Only works before a call is started.
- return -1;
- }
-
- // Most of places NetEq calls are not within AcmReceiver's critical section to
- // improve performance. Here, this call has to be placed before the following
- // block, therefore, we keep it inside critical section. Otherwise, we have to
- // release |neteq_crit_sect_| and acquire it again, which seems an overkill.
- if (!neteq_->SetMinimumDelay(delay_ms))
- return -1;
-
- const int kLatePacketThreshold = 5;
- av_sync_ = true;
- initial_delay_manager_.reset(new InitialDelayManager(delay_ms,
- kLatePacketThreshold));
- missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
- late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
- return 0;
-}
-
int AcmReceiver::SetMaximumDelay(int delay_ms) {
if (neteq_->SetMaximumDelay(delay_ms))
return 0;
@@ -207,9 +166,6 @@
const uint8_t* incoming_payload,
size_t length_payload) {
uint32_t receive_timestamp = 0;
- InitialDelayManager::PacketType packet_type =
- InitialDelayManager::kUndefinedPacket;
- bool new_codec = false;
const RTPHeader* header = &rtp_header.header; // Just a shorthand.
{
@@ -225,45 +181,19 @@
const int sample_rate_hz = ACMCodecDB::CodecFreq(decoder->acm_codec_id);
receive_timestamp = NowInTimestamp(sample_rate_hz);
- if (IsCng(decoder->acm_codec_id)) {
- // If this is a CNG while the audio codec is not mono skip pushing in
- // packets into NetEq.
- if (last_audio_decoder_ && last_audio_decoder_->channels > 1)
+ // If this is a CNG while the audio codec is not mono, skip pushing in
+ // packets into NetEq.
+ if (IsCng(decoder->acm_codec_id) && last_audio_decoder_ &&
+ last_audio_decoder_->channels > 1)
return 0;
- packet_type = InitialDelayManager::kCngPacket;
- } else if (decoder->acm_codec_id ==
- *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
- packet_type = InitialDelayManager::kAvtPacket;
- } else {
- if (decoder != last_audio_decoder_) {
- // This is either the first audio packet or send codec is changed.
- // Therefore, either NetEq buffer is empty or will be flushed when this
- // packet is inserted.
- new_codec = true;
- last_audio_decoder_ = decoder;
- }
- packet_type = InitialDelayManager::kAudioPacket;
+ if (!IsCng(decoder->acm_codec_id) &&
+ decoder->acm_codec_id !=
+ *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
+ last_audio_decoder_ = decoder;
}
- if (av_sync_) {
- assert(initial_delay_manager_.get());
- assert(missing_packets_sync_stream_.get());
- // This updates |initial_delay_manager_| and specifies an stream of
- // sync-packets, if required to be inserted. We insert the sync-packets
- // when AcmReceiver lock is released and |decoder_lock_| is acquired.
- initial_delay_manager_->UpdateLastReceivedPacket(
- rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
- missing_packets_sync_stream_.get());
- }
} // |crit_sect_| is released.
- // If |missing_packets_sync_stream_| is allocated then we are in AV-sync and
- // we may need to insert sync-packets. We don't check |av_sync_| as we are
- // outside AcmReceiver's critical section.
- if (missing_packets_sync_stream_.get()) {
- InsertStreamOfSyncPackets(missing_packets_sync_stream_.get());
- }
-
if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
receive_timestamp) < 0) {
LOG(LERROR) << "AcmReceiver::InsertPacket "
@@ -278,29 +208,6 @@
enum NetEqOutputType type;
size_t samples_per_channel;
int num_channels;
- bool return_silence = false;
-
- {
- // Accessing members, take the lock.
- CriticalSectionScoped lock(crit_sect_.get());
-
- if (av_sync_) {
- assert(initial_delay_manager_.get());
- assert(late_packets_sync_stream_.get());
- return_silence = GetSilence(desired_freq_hz, audio_frame);
- uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_);
- initial_delay_manager_->LatePackets(timestamp_now,
- late_packets_sync_stream_.get());
- }
- }
-
- // If |late_packets_sync_stream_| is allocated then we have been in AV-sync
- // mode and we might have to insert sync-packets.
- if (late_packets_sync_stream_.get()) {
- InsertStreamOfSyncPackets(late_packets_sync_stream_.get());
- if (return_silence) // Silence generated, don't pull from NetEq.
- return 0;
- }
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
@@ -519,12 +426,6 @@
}
bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
- if (av_sync_) {
- assert(initial_delay_manager_.get());
- if (initial_delay_manager_->buffering()) {
- return initial_delay_manager_->GetPlayoutTimestamp(timestamp);
- }
- }
return neteq_->GetPlayoutTimestamp(timestamp);
}
@@ -602,64 +503,10 @@
}
void AcmReceiver::ResetInitialDelay() {
- {
- CriticalSectionScoped lock(crit_sect_.get());
- av_sync_ = false;
- initial_delay_manager_.reset(NULL);
- missing_packets_sync_stream_.reset(NULL);
- late_packets_sync_stream_.reset(NULL);
- }
neteq_->SetMinimumDelay(0);
// TODO(turajs): Should NetEq Buffer be flushed?
}
-// This function is called within critical section, no need to acquire a lock.
-bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
- assert(av_sync_);
- assert(initial_delay_manager_.get());
- if (!initial_delay_manager_->buffering()) {
- return false;
- }
-
- // We stop accumulating packets, if the number of packets or the total size
- // exceeds a threshold.
- int num_packets;
- int max_num_packets;
- const float kBufferingThresholdScale = 0.9f;
- neteq_->PacketBufferStatistics(&num_packets, &max_num_packets);
- if (num_packets > max_num_packets * kBufferingThresholdScale) {
- initial_delay_manager_->DisableBuffering();
- return false;
- }
-
- // Update statistics.
- call_stats_.DecodedBySilenceGenerator();
-
- // Set the values if already got a packet, otherwise set to default values.
- if (last_audio_decoder_) {
- current_sample_rate_hz_ =
- ACMCodecDB::database_[last_audio_decoder_->acm_codec_id].plfreq;
- frame->num_channels_ = last_audio_decoder_->channels;
- } else {
- frame->num_channels_ = 1;
- }
-
- // Set the audio frame's sampling frequency.
- if (desired_sample_rate_hz > 0) {
- frame->sample_rate_hz_ = desired_sample_rate_hz;
- } else {
- frame->sample_rate_hz_ = current_sample_rate_hz_;
- }
-
- frame->samples_per_channel_ =
- static_cast<size_t>(frame->sample_rate_hz_ / 100); // Always 10 ms.
- frame->speech_type_ = AudioFrame::kCNG;
- frame->vad_activity_ = AudioFrame::kVadPassive;
- size_t samples = frame->samples_per_channel_ * frame->num_channels_;
- memset(frame->data_, 0, samples * sizeof(int16_t));
- return true;
-}
-
const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
const RTPHeader& rtp_header,
const uint8_t* payload) const {
@@ -687,23 +534,6 @@
(decoder_sampling_rate / 1000) * now_in_ms);
}
-// This function only interacts with |neteq_|, therefore, it does not have to
-// be within critical section of AcmReceiver. It is inserting packets
-// into NetEq, so we call it when |decode_lock_| is acquired. However, this is
-// not essential as sync-packets do not interact with codecs (especially BWE).
-void AcmReceiver::InsertStreamOfSyncPackets(
- InitialDelayManager::SyncStream* sync_stream) {
- assert(sync_stream);
- assert(av_sync_);
- for (int n = 0; n < sync_stream->num_sync_packets; ++n) {
- neteq_->InsertSyncPacket(sync_stream->rtp_info,
- sync_stream->receive_timestamp);
- ++sync_stream->rtp_info.header.sequenceNumber;
- sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
- sync_stream->receive_timestamp += sync_stream->timestamp_step;
- }
-}
-
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
CriticalSectionScoped lock(crit_sect_.get());
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index aee9154..9e812b2 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -151,19 +151,6 @@
int LeastRequiredDelayMs() const;
//
- // Sets an initial delay of |delay_ms| milliseconds. This introduces a playout
- // delay. Silence (zero signal) is played out until equivalent of |delay_ms|
- // millisecond of audio is buffered. Then, NetEq maintains the delay.
- //
- // Input:
- // - delay_ms : initial delay in milliseconds.
- //
- // Return value : 0 if OK.
- // <0 if NetEq returned an error.
- //
- int SetInitialDelay(int delay_ms);
-
- //
// Resets the initial delay to zero.
//
void ResetInitialDelay();
@@ -291,19 +278,12 @@
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
private:
- bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame)
- EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
- int GetNumSyncPacketToInsert(uint16_t received_squence_number);
-
const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
const uint8_t* payload) const
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
- void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
-
rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
int id_; // TODO(henrik.lundin) Make const.
const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
@@ -321,19 +301,6 @@
bool vad_enabled_;
Clock* clock_; // TODO(henrik.lundin) Make const if possible.
bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
-
- // Indicates if a non-zero initial delay is set, and the receiver is in
- // AV-sync mode.
- bool av_sync_;
- rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_;
-
- // The following are defined as members to avoid creating them in every
- // iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket().
- // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
- // member variables are allocated only when we AV-sync is enabled, i.e.
- // initial delay is set.
- rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
- rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
};
} // namespace acm2
