Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"
This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index afc15bf..4630448 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -101,7 +101,6 @@
reset_decoder_(false),
ssrc_(0),
first_packet_(true),
- playout_mode_(config.playout_mode),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false),
enable_muted_state_(config.enable_muted_state),
@@ -110,7 +109,8 @@
tick_timer_.get()),
speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
10, // Report once every 10 s.
- tick_timer_.get()) {
+ tick_timer_.get()),
+ no_time_stretching_(config.for_test_no_time_stretching) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@@ -358,23 +358,6 @@
return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
}
-// Deprecated.
-// TODO(henrik.lundin) Delete.
-void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
- rtc::CritScope lock(&crit_sect_);
- if (mode != playout_mode_) {
- playout_mode_ = mode;
- CreateDecisionLogic();
- }
-}
-
-// Deprecated.
-// TODO(henrik.lundin) Delete.
-NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
- rtc::CritScope lock(&crit_sect_);
- return playout_mode_;
-}
-
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
rtc::CritScope lock(&crit_sect_);
assert(decoder_database_.get());
@@ -937,33 +920,6 @@
return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
- case kAlternativePlc: {
- // TODO(hlundin): Write test for this.
- DoAlternativePlc(false);
- break;
- }
- case kAlternativePlcIncreaseTimestamp: {
- // TODO(hlundin): Write test for this.
- DoAlternativePlc(true);
- break;
- }
- case kAudioRepetitionIncreaseTimestamp: {
- // TODO(hlundin): Write test for this.
- sync_buffer_->IncreaseEndTimestamp(
- static_cast<uint32_t>(output_size_samples_));
- // Skipping break on purpose. Execution should move on into the
- // next case.
- RTC_FALLTHROUGH();
- }
- case kAudioRepetition: {
- // TODO(hlundin): Write test for this.
- // Copy last |output_size_samples_| from |sync_buffer_| to
- // |algorithm_buffer|.
- algorithm_buffer_->PushBackFromIndex(
- *sync_buffer_, sync_buffer_->Size() - output_size_samples_);
- expand_->Reset();
- break;
- }
case kUndefined: {
RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
assert(false); // This should not happen.
@@ -1291,10 +1247,7 @@
// Get packets from buffer.
int extracted_samples = 0;
- if (packet && *operation != kAlternativePlc &&
- *operation != kAlternativePlcIncreaseTimestamp &&
- *operation != kAudioRepetition &&
- *operation != kAudioRepetitionIncreaseTimestamp) {
+ if (packet) {
sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
if (decision_logic_->CngOff()) {
// Adjustment of timestamp only corresponds to an actual packet loss
@@ -1883,29 +1836,6 @@
return 0;
}
-void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
- AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
- size_t length;
- if (decoder && decoder->HasDecodePlc()) {
- // Use the decoder's packet-loss concealment.
- // TODO(hlundin): Will probably need a longer buffer for multi-channel.
- int16_t decoded_buffer[kMaxFrameSize];
- length = decoder->DecodePlc(1, decoded_buffer);
- if (length > 0)
- algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
- } else {
- // Do simple zero-stuffing.
- length = output_size_samples_;
- algorithm_buffer_->Zeros(length);
- // By not advancing the timestamp, NetEq inserts samples.
- stats_.AddZeros(length);
- }
- if (increase_timestamp) {
- sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
- }
- expand_->Reset();
-}
-
int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const {
@@ -2130,8 +2060,8 @@
void NetEqImpl::CreateDecisionLogic() {
decision_logic_.reset(DecisionLogic::Create(
- fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
- *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
- tick_timer_.get()));
+ fs_hz_, output_size_samples_, no_time_stretching_,
+ decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(),
+ buffer_level_filter_.get(), tick_timer_.get()));
}
} // namespace webrtc