modules/audio_coding/acm2/acm_receiver.cc - src - Git at Google

 /*
  *  Copyright (c) 2013 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.
  */

 #include "modules/audio_coding/acm2/acm_receiver.h"

 #include <stdlib.h>
 #include <string.h>

 #include <cstdint>
 #include <vector>

 #include "absl/strings/match.h"
 #include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/neteq/neteq.h"
 #include "modules/audio_coding/acm2/acm_resampler.h"
 #include "modules/audio_coding/acm2/call_statistics.h"
 #include "modules/audio_coding/neteq/default_neteq_factory.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/audio_format_to_string.h"
 #include "system_wrappers/include/clock.h"

 namespace webrtc {

 namespace acm2 {

 namespace {

 std::unique_ptr<NetEq> CreateNetEq(
     NetEqFactory* neteq_factory,
     const NetEq::Config& config,
     Clock* clock,
     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
   if (neteq_factory) {
     return neteq_factory->CreateNetEq(config, decoder_factory, clock);
   }
   return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
 }

 }  // namespace

 AcmReceiver::Config::Config(
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
     : clock(*Clock::GetRealTimeClock()), decoder_factory(decoder_factory) {}

 AcmReceiver::Config::Config(const Config&) = default;
 AcmReceiver::Config::~Config() = default;

 AcmReceiver::AcmReceiver(const Config& config)
     : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       neteq_(CreateNetEq(config.neteq_factory,
                          config.neteq_config,
                          &config.clock,
                          config.decoder_factory)),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
   memset(last_audio_buffer_.get(), 0,
          sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
 }

 AcmReceiver::~AcmReceiver() = default;

 int AcmReceiver::SetMinimumDelay(int delay_ms) {
   if (neteq_->SetMinimumDelay(delay_ms))
     return 0;
   RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
   return -1;
 }

 int AcmReceiver::SetMaximumDelay(int delay_ms) {
   if (neteq_->SetMaximumDelay(delay_ms))
     return 0;
   RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
   return -1;
 }

 bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) {
   return neteq_->SetBaseMinimumDelayMs(delay_ms);
 }

 int AcmReceiver::GetBaseMinimumDelayMs() const {
   return neteq_->GetBaseMinimumDelayMs();
 }

 absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
   MutexLock lock(&mutex_);
   if (!last_decoder_) {
     return absl::nullopt;
   }
   return last_decoder_->sample_rate_hz;
 }

 int AcmReceiver::last_output_sample_rate_hz() const {
   return neteq_->last_output_sample_rate_hz();
 }

 int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
                               rtc::ArrayView<const uint8_t> incoming_payload) {
   if (incoming_payload.empty()) {
     neteq_->InsertEmptyPacket(rtp_header);
     return 0;
   }

   int payload_type = rtp_header.payloadType;
   auto format = neteq_->GetDecoderFormat(payload_type);
   if (format && absl::EqualsIgnoreCase(format->sdp_format.name, "red")) {
     // This is a RED packet. Get the format of the audio codec.
     payload_type = incoming_payload[0] & 0x7f;
     format = neteq_->GetDecoderFormat(payload_type);
   }
   if (!format) {
     RTC_LOG_F(LS_ERROR) << "Payload-type " << payload_type
                         << " is not registered.";
     return -1;
   }

   {
     MutexLock lock(&mutex_);
     if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
       if (last_decoder_ && last_decoder_->num_channels > 1) {
         // This is a CNG and the audio codec is not mono, so skip pushing in
         // packets into NetEq.
         return 0;
       }
     } else {
       last_decoder_ = DecoderInfo{/*payload_type=*/payload_type,
                                   /*sample_rate_hz=*/format->sample_rate_hz,
                                   /*num_channels=*/format->num_channels,
                                   /*sdp_format=*/std::move(format->sdp_format)};
     }
   }  // `mutex_` is released.

   if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
     RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
                       << static_cast<int>(rtp_header.payloadType)
                       << " Failed to insert packet";
     return -1;
   }
   return 0;
 }

 int AcmReceiver::GetAudio(int desired_freq_hz,
                           AudioFrame* audio_frame,
                           bool* muted) {
   int current_sample_rate_hz = 0;
   if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
       NetEq::kOK) {
     RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }

   RTC_DCHECK_NE(current_sample_rate_hz, 0);

   // Update if resampling is required.
   const bool need_resampling =
       (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);

   // Accessing members, take the lock.
   MutexLock lock(&mutex_);
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
     int samples_per_channel_int = resampler_.Resample10Msec(
         last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
         audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
         temp_output);
     if (samples_per_channel_int < 0) {
       RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
                            "Resampling last_audio_buffer_ failed.";
       return -1;
     }
   }

   // TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
   // rate from NetEq changes.
   if (need_resampling) {
     // TODO(yujo): handle this more efficiently for muted frames.
     int samples_per_channel_int = resampler_.Resample10Msec(
         audio_frame->data(), current_sample_rate_hz, desired_freq_hz,
         audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
         audio_frame->mutable_data());
     if (samples_per_channel_int < 0) {
       RTC_LOG(LS_ERROR)
           << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
       return -1;
     }
     audio_frame->samples_per_channel_ =
         static_cast<size_t>(samples_per_channel_int);
     audio_frame->sample_rate_hz_ = desired_freq_hz;
     RTC_DCHECK_EQ(
         audio_frame->sample_rate_hz_,
         rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
     resampled_last_output_frame_ = true;
   } else {
     resampled_last_output_frame_ = false;
     // We might end up here ONLY if codec is changed.
   }

   // Store current audio in `last_audio_buffer_` for next time.
   memcpy(last_audio_buffer_.get(), audio_frame->data(),
          sizeof(int16_t) * audio_frame->samples_per_channel_ *
              audio_frame->num_channels_);

   call_stats_.DecodedByNetEq(audio_frame->speech_type_, audio_frame->muted());
   return 0;
 }

 void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
   neteq_->SetCodecs(codecs);
 }

 void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }

 void AcmReceiver::RemoveAllCodecs() {
   MutexLock lock(&mutex_);
   neteq_->RemoveAllPayloadTypes();
   last_decoder_ = absl::nullopt;
 }

 absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
   return neteq_->GetPlayoutTimestamp();
 }

 int AcmReceiver::FilteredCurrentDelayMs() const {
   return neteq_->FilteredCurrentDelayMs();
 }

 int AcmReceiver::TargetDelayMs() const {
   return neteq_->TargetDelayMs();
 }

 absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
     const {
   MutexLock lock(&mutex_);
   if (!last_decoder_) {
     return absl::nullopt;
   }
   RTC_DCHECK_NE(-1, last_decoder_->payload_type);
   return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
 }

 void AcmReceiver::GetNetworkStatistics(
     NetworkStatistics* acm_stat,
     bool get_and_clear_legacy_stats /* = true */) const {
   NetEqNetworkStatistics neteq_stat;
   if (get_and_clear_legacy_stats) {
     // NetEq function always returns zero, so we don't check the return value.
     neteq_->NetworkStatistics(&neteq_stat);

     acm_stat->currentExpandRate = neteq_stat.expand_rate;
     acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
     acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
     acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
     acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
     acm_stat->currentSecondaryDiscardedRate =
         neteq_stat.secondary_discarded_rate;
     acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
     acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
   } else {
     neteq_stat = neteq_->CurrentNetworkStatistics();
     acm_stat->currentExpandRate = 0;
     acm_stat->currentSpeechExpandRate = 0;
     acm_stat->currentPreemptiveRate = 0;
     acm_stat->currentAccelerateRate = 0;
     acm_stat->currentSecondaryDecodedRate = 0;
     acm_stat->currentSecondaryDiscardedRate = 0;
     acm_stat->meanWaitingTimeMs = -1;
     acm_stat->maxWaitingTimeMs = 1;
   }
   acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
   acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
   acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;

   NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
   acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
   acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
   acm_stat->silentConcealedSamples =
       neteq_lifetime_stat.silent_concealed_samples;
   acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
   acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
   acm_stat->jitterBufferTargetDelayMs =
       neteq_lifetime_stat.jitter_buffer_target_delay_ms;
   acm_stat->jitterBufferMinimumDelayMs =
       neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
   acm_stat->jitterBufferEmittedCount =
       neteq_lifetime_stat.jitter_buffer_emitted_count;
   acm_stat->delayedPacketOutageSamples =
       neteq_lifetime_stat.delayed_packet_outage_samples;
   acm_stat->relativePacketArrivalDelayMs =
       neteq_lifetime_stat.relative_packet_arrival_delay_ms;
   acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
   acm_stat->totalInterruptionDurationMs =
       neteq_lifetime_stat.total_interruption_duration_ms;
   acm_stat->insertedSamplesForDeceleration =
       neteq_lifetime_stat.inserted_samples_for_deceleration;
   acm_stat->removedSamplesForAcceleration =
       neteq_lifetime_stat.removed_samples_for_acceleration;
   acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
   acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
   acm_stat->packetsDiscarded = neteq_lifetime_stat.packets_discarded;

   NetEqOperationsAndState neteq_operations_and_state =
       neteq_->GetOperationsAndState();
   acm_stat->packetBufferFlushes =
       neteq_operations_and_state.packet_buffer_flushes;
 }

 int AcmReceiver::EnableNack(size_t max_nack_list_size) {
   neteq_->EnableNack(max_nack_list_size);
   return 0;
 }

 void AcmReceiver::DisableNack() {
   neteq_->DisableNack();
 }

 std::vector<uint16_t> AcmReceiver::GetNackList(
     int64_t round_trip_time_ms) const {
   return neteq_->GetNackList(round_trip_time_ms);
 }

 void AcmReceiver::ResetInitialDelay() {
   neteq_->SetMinimumDelay(0);
   // TODO(turajs): Should NetEq Buffer be flushed?
 }

 uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
   // Down-cast the time to (32-6)-bit since we only care about
   // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
   // We masked 6 most significant bits of 32-bit so there is no overflow in
   // the conversion from milliseconds to timestamp.
   const uint32_t now_in_ms =
       static_cast<uint32_t>(clock_.TimeInMilliseconds() & 0x03ffffff);
   return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
 }

 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   MutexLock lock(&mutex_);
   *stats = call_stats_.GetDecodingStatistics();
 }

 }  // namespace acm2

 }  // namespace webrtc
	/*
	* Copyright (c) 2013 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.
	*/

	#include "modules/audio_coding/acm2/acm_receiver.h"

	#include <stdlib.h>
	#include <string.h>

	#include <cstdint>
	#include <vector>

	#include "absl/strings/match.h"
	#include "api/audio/audio_frame.h"
	#include "api/audio_codecs/audio_decoder.h"
	#include "api/neteq/neteq.h"
	#include "modules/audio_coding/acm2/acm_resampler.h"
	#include "modules/audio_coding/acm2/call_statistics.h"
	#include "modules/audio_coding/neteq/default_neteq_factory.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/numerics/safe_conversions.h"
	#include "rtc_base/strings/audio_format_to_string.h"
	#include "system_wrappers/include/clock.h"

	namespace webrtc {

	namespace acm2 {

	namespace {

	std::unique_ptr<NetEq> CreateNetEq(
	NetEqFactory* neteq_factory,
	const NetEq::Config& config,
	Clock* clock,
	const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
	if (neteq_factory) {
	return neteq_factory->CreateNetEq(config, decoder_factory, clock);
	}
	return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
	}

	} // namespace

	AcmReceiver::Config::Config(
	rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
	: clock(*Clock::GetRealTimeClock()), decoder_factory(decoder_factory) {}

	AcmReceiver::Config::Config(const Config&) = default;
	AcmReceiver::Config::~Config() = default;

	AcmReceiver::AcmReceiver(const Config& config)
	: last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
	neteq_(CreateNetEq(config.neteq_factory,
	config.neteq_config,
	&config.clock,
	config.decoder_factory)),
	clock_(config.clock),
	resampled_last_output_frame_(true) {
	memset(last_audio_buffer_.get(), 0,
	sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
	}

	AcmReceiver::~AcmReceiver() = default;

	int AcmReceiver::SetMinimumDelay(int delay_ms) {
	if (neteq_->SetMinimumDelay(delay_ms))
	return 0;
	RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
	return -1;
	}

	int AcmReceiver::SetMaximumDelay(int delay_ms) {
	if (neteq_->SetMaximumDelay(delay_ms))
	return 0;
	RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
	return -1;
	}

	bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) {
	return neteq_->SetBaseMinimumDelayMs(delay_ms);
	}

	int AcmReceiver::GetBaseMinimumDelayMs() const {
	return neteq_->GetBaseMinimumDelayMs();
	}

	absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
	MutexLock lock(&mutex_);
	if (!last_decoder_) {
	return absl::nullopt;
	}
	return last_decoder_->sample_rate_hz;
	}

	int AcmReceiver::last_output_sample_rate_hz() const {
	return neteq_->last_output_sample_rate_hz();
	}

	int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
	rtc::ArrayView<const uint8_t> incoming_payload) {
	if (incoming_payload.empty()) {
	neteq_->InsertEmptyPacket(rtp_header);
	return 0;
	}

	int payload_type = rtp_header.payloadType;
	auto format = neteq_->GetDecoderFormat(payload_type);
	if (format && absl::EqualsIgnoreCase(format->sdp_format.name, "red")) {
	// This is a RED packet. Get the format of the audio codec.
	payload_type = incoming_payload[0] & 0x7f;
	format = neteq_->GetDecoderFormat(payload_type);
	}
	if (!format) {
	RTC_LOG_F(LS_ERROR) << "Payload-type " << payload_type
	<< " is not registered.";
	return -1;
	}

	{
	MutexLock lock(&mutex_);
	if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
	if (last_decoder_ && last_decoder_->num_channels > 1) {
	// This is a CNG and the audio codec is not mono, so skip pushing in
	// packets into NetEq.
	return 0;
	}
	} else {
	last_decoder_ = DecoderInfo{/payload_type=/payload_type,
	/sample_rate_hz=/format->sample_rate_hz,
	/num_channels=/format->num_channels,
	/sdp_format=/std::move(format->sdp_format)};
	}
	} // `mutex_` is released.

	if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
	RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
	<< static_cast<int>(rtp_header.payloadType)
	<< " Failed to insert packet";
	return -1;
	}
	return 0;
	}

	int AcmReceiver::GetAudio(int desired_freq_hz,
	AudioFrame* audio_frame,
	bool* muted) {
	int current_sample_rate_hz = 0;
	if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
	NetEq::kOK) {
	RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
	return -1;
	}

	RTC_DCHECK_NE(current_sample_rate_hz, 0);

	// Update if resampling is required.
	const bool need_resampling =
	(desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);

	// Accessing members, take the lock.
	MutexLock lock(&mutex_);
	if (need_resampling && !resampled_last_output_frame_) {
	// Prime the resampler with the last frame.
	int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
	int samples_per_channel_int = resampler_.Resample10Msec(
	last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
	audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
	temp_output);
	if (samples_per_channel_int < 0) {
	RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
	"Resampling last_audio_buffer_ failed.";
	return -1;
	}
	}

	// TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
	// rate from NetEq changes.
	if (need_resampling) {
	// TODO(yujo): handle this more efficiently for muted frames.
	int samples_per_channel_int = resampler_.Resample10Msec(
	audio_frame->data(), current_sample_rate_hz, desired_freq_hz,
	audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
	audio_frame->mutable_data());
	if (samples_per_channel_int < 0) {
	RTC_LOG(LS_ERROR)
	<< "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
	return -1;
	}
	audio_frame->samples_per_channel_ =
	static_cast<size_t>(samples_per_channel_int);
	audio_frame->sample_rate_hz_ = desired_freq_hz;
	RTC_DCHECK_EQ(
	audio_frame->sample_rate_hz_,
	rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
	resampled_last_output_frame_ = true;
	} else {
	resampled_last_output_frame_ = false;
	// We might end up here ONLY if codec is changed.
	}

	// Store current audio in `last_audio_buffer_` for next time.
	memcpy(last_audio_buffer_.get(), audio_frame->data(),
	sizeof(int16_t) * audio_frame->samples_per_channel_ *
	audio_frame->num_channels_);

	call_stats_.DecodedByNetEq(audio_frame->speech_type_, audio_frame->muted());
	return 0;
	}

	void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
	neteq_->SetCodecs(codecs);
	}

	void AcmReceiver::FlushBuffers() {
	neteq_->FlushBuffers();
	}

	void AcmReceiver::RemoveAllCodecs() {
	MutexLock lock(&mutex_);
	neteq_->RemoveAllPayloadTypes();
	last_decoder_ = absl::nullopt;
	}

	absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
	return neteq_->GetPlayoutTimestamp();
	}

	int AcmReceiver::FilteredCurrentDelayMs() const {
	return neteq_->FilteredCurrentDelayMs();
	}

	int AcmReceiver::TargetDelayMs() const {
	return neteq_->TargetDelayMs();
	}

	absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
	const {
	MutexLock lock(&mutex_);
	if (!last_decoder_) {
	return absl::nullopt;
	}
	RTC_DCHECK_NE(-1, last_decoder_->payload_type);
	return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
	}

	void AcmReceiver::GetNetworkStatistics(
	NetworkStatistics* acm_stat,
	bool get_and_clear_legacy_stats /* = true */) const {
	NetEqNetworkStatistics neteq_stat;
	if (get_and_clear_legacy_stats) {
	// NetEq function always returns zero, so we don't check the return value.
	neteq_->NetworkStatistics(&neteq_stat);

	acm_stat->currentExpandRate = neteq_stat.expand_rate;
	acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
	acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
	acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
	acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
	acm_stat->currentSecondaryDiscardedRate =
	neteq_stat.secondary_discarded_rate;
	acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
	acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
	} else {
	neteq_stat = neteq_->CurrentNetworkStatistics();
	acm_stat->currentExpandRate = 0;
	acm_stat->currentSpeechExpandRate = 0;
	acm_stat->currentPreemptiveRate = 0;
	acm_stat->currentAccelerateRate = 0;
	acm_stat->currentSecondaryDecodedRate = 0;
	acm_stat->currentSecondaryDiscardedRate = 0;
	acm_stat->meanWaitingTimeMs = -1;
	acm_stat->maxWaitingTimeMs = 1;
	}
	acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
	acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
	acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;

	NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
	acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
	acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
	acm_stat->silentConcealedSamples =
	neteq_lifetime_stat.silent_concealed_samples;
	acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
	acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
	acm_stat->jitterBufferTargetDelayMs =
	neteq_lifetime_stat.jitter_buffer_target_delay_ms;
	acm_stat->jitterBufferMinimumDelayMs =
	neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
	acm_stat->jitterBufferEmittedCount =
	neteq_lifetime_stat.jitter_buffer_emitted_count;
	acm_stat->delayedPacketOutageSamples =
	neteq_lifetime_stat.delayed_packet_outage_samples;
	acm_stat->relativePacketArrivalDelayMs =
	neteq_lifetime_stat.relative_packet_arrival_delay_ms;
	acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
	acm_stat->totalInterruptionDurationMs =
	neteq_lifetime_stat.total_interruption_duration_ms;
	acm_stat->insertedSamplesForDeceleration =
	neteq_lifetime_stat.inserted_samples_for_deceleration;
	acm_stat->removedSamplesForAcceleration =
	neteq_lifetime_stat.removed_samples_for_acceleration;
	acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
	acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
	acm_stat->packetsDiscarded = neteq_lifetime_stat.packets_discarded;

	NetEqOperationsAndState neteq_operations_and_state =
	neteq_->GetOperationsAndState();
	acm_stat->packetBufferFlushes =
	neteq_operations_and_state.packet_buffer_flushes;
	}

	int AcmReceiver::EnableNack(size_t max_nack_list_size) {
	neteq_->EnableNack(max_nack_list_size);
	return 0;
	}

	void AcmReceiver::DisableNack() {
	neteq_->DisableNack();
	}

	std::vector<uint16_t> AcmReceiver::GetNackList(
	int64_t round_trip_time_ms) const {
	return neteq_->GetNackList(round_trip_time_ms);
	}

	void AcmReceiver::ResetInitialDelay() {
	neteq_->SetMinimumDelay(0);
	// TODO(turajs): Should NetEq Buffer be flushed?
	}

	uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
	// Down-cast the time to (32-6)-bit since we only care about
	// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
	// We masked 6 most significant bits of 32-bit so there is no overflow in
	// the conversion from milliseconds to timestamp.
	const uint32_t now_in_ms =
	static_cast<uint32_t>(clock_.TimeInMilliseconds() & 0x03ffffff);
	return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
	}

	void AcmReceiver::GetDecodingCallStatistics(
	AudioDecodingCallStats* stats) const {
	MutexLock lock(&mutex_);
	*stats = call_stats_.GetDecodingStatistics();
	}

	} // namespace acm2

	} // namespace webrtc