Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(403)

Unified Diff: webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
deleted file mode 100644
index 6c2893336a8290388800d8e4b24247c920f358e8..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ /dev/null
@@ -1,540 +0,0 @@
-/*
- * 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 "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-
-#include <stdlib.h> // malloc
-
-#include <algorithm> // sort
-#include <vector>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/format_macros.h"
-#include "webrtc/base/logging.h"
-#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_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-namespace {
-
-// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
-// before the call to this function.
-void SetAudioFrameActivityAndType(bool vad_enabled,
- NetEqOutputType type,
- AudioFrame* audio_frame) {
- if (vad_enabled) {
- switch (type) {
- case kOutputNormal: {
- audio_frame->vad_activity_ = AudioFrame::kVadActive;
- audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
- break;
- }
- case kOutputVADPassive: {
- audio_frame->vad_activity_ = AudioFrame::kVadPassive;
- audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
- break;
- }
- case kOutputCNG: {
- audio_frame->vad_activity_ = AudioFrame::kVadPassive;
- audio_frame->speech_type_ = AudioFrame::kCNG;
- break;
- }
- case kOutputPLC: {
- // Don't change |audio_frame->vad_activity_|, it should be the same as
- // |previous_audio_activity_|.
- audio_frame->speech_type_ = AudioFrame::kPLC;
- break;
- }
- case kOutputPLCtoCNG: {
- audio_frame->vad_activity_ = AudioFrame::kVadPassive;
- audio_frame->speech_type_ = AudioFrame::kPLCCNG;
- break;
- }
- default:
- assert(false);
- }
- } else {
- // Always return kVadUnknown when receive VAD is inactive
- audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
- switch (type) {
- case kOutputNormal: {
- audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
- break;
- }
- case kOutputCNG: {
- audio_frame->speech_type_ = AudioFrame::kCNG;
- break;
- }
- case kOutputPLC: {
- audio_frame->speech_type_ = AudioFrame::kPLC;
- break;
- }
- case kOutputPLCtoCNG: {
- audio_frame->speech_type_ = AudioFrame::kPLCCNG;
- break;
- }
- case kOutputVADPassive: {
- // Normally, we should no get any VAD decision if post-decoding VAD is
- // not active. However, if post-decoding VAD has been active then
- // disabled, we might be here for couple of frames.
- audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
- LOG(WARNING) << "Post-decoding VAD is disabled but output is "
- << "labeled VAD-passive";
- break;
- }
- default:
- assert(false);
- }
- }
-}
-
-// Is the given codec a CNG codec?
-// TODO(kwiberg): Move to RentACodec.
-bool IsCng(int codec_id) {
- auto i = RentACodec::CodecIdFromIndex(codec_id);
- return (i && (*i == RentACodec::CodecId::kCNNB ||
- *i == RentACodec::CodecId::kCNWB ||
- *i == RentACodec::CodecId::kCNSWB ||
- *i == RentACodec::CodecId::kCNFB));
-}
-
-} // namespace
-
-AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- id_(config.id),
- last_audio_decoder_(nullptr),
- previous_audio_activity_(AudioFrame::kVadPassive),
- audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
- last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
- neteq_(NetEq::Create(config.neteq_config)),
- vad_enabled_(config.neteq_config.enable_post_decode_vad),
- clock_(config.clock),
- resampled_last_output_frame_(true) {
- assert(clock_);
- memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
- memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
-}
-
-AcmReceiver::~AcmReceiver() {
- delete neteq_;
-}
-
-int AcmReceiver::SetMinimumDelay(int delay_ms) {
- if (neteq_->SetMinimumDelay(delay_ms))
- return 0;
- LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
- return -1;
-}
-
-int AcmReceiver::SetMaximumDelay(int delay_ms) {
- if (neteq_->SetMaximumDelay(delay_ms))
- return 0;
- LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
- return -1;
-}
-
-int AcmReceiver::LeastRequiredDelayMs() const {
- return neteq_->LeastRequiredDelayMs();
-}
-
-rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
- CriticalSectionScoped lock(crit_sect_.get());
- return last_packet_sample_rate_hz_;
-}
-
-int AcmReceiver::last_output_sample_rate_hz() const {
- return neteq_->last_output_sample_rate_hz();
-}
-
-int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> incoming_payload) {
- uint32_t receive_timestamp = 0;
- const RTPHeader* header = &rtp_header.header; // Just a shorthand.
-
- {
- CriticalSectionScoped lock(crit_sect_.get());
-
- const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
- if (!decoder) {
- LOG_F(LS_ERROR) << "Payload-type "
- << static_cast<int>(header->payloadType)
- << " is not registered.";
- return -1;
- }
- const int sample_rate_hz = [&decoder] {
- const auto ci = RentACodec::CodecIdFromIndex(decoder->acm_codec_id);
- return ci ? RentACodec::CodecInstById(*ci)->plfreq : -1;
- }();
- receive_timestamp = NowInTimestamp(sample_rate_hz);
-
- // 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;
- if (!IsCng(decoder->acm_codec_id) &&
- decoder->acm_codec_id !=
- *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
- last_audio_decoder_ = decoder;
- last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
- }
-
- } // |crit_sect_| is released.
-
- if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
- 0) {
- LOG(LERROR) << "AcmReceiver::InsertPacket "
- << static_cast<int>(header->payloadType)
- << " Failed to insert packet";
- return -1;
- }
- return 0;
-}
-
-int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
- enum NetEqOutputType type;
- size_t samples_per_channel;
- int num_channels;
-
- // Accessing members, take the lock.
- CriticalSectionScoped lock(crit_sect_.get());
-
- // Always write the output to |audio_buffer_| first.
- if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
- audio_buffer_.get(),
- &samples_per_channel,
- &num_channels,
- &type) != NetEq::kOK) {
- LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
- return -1;
- }
-
- const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
-
- // Update if resampling is required.
- const bool need_resampling =
- (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
-
- 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,
- num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
- if (samples_per_channel_int < 0) {
- LOG(LERROR) << "AcmReceiver::GetAudio - "
- "Resampling last_audio_buffer_ failed.";
- return -1;
- }
- samples_per_channel = static_cast<size_t>(samples_per_channel_int);
- }
-
- // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
- // through resampling, or through straight memcpy.
- // TODO(henrik.lundin) Glitches in the output may appear if the output rate
- // from NetEq changes. See WebRTC issue 3923.
- if (need_resampling) {
- int samples_per_channel_int = resampler_.Resample10Msec(
- audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
- num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
- if (samples_per_channel_int < 0) {
- LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
- return -1;
- }
- samples_per_channel = static_cast<size_t>(samples_per_channel_int);
- resampled_last_output_frame_ = true;
- } else {
- resampled_last_output_frame_ = false;
- // We might end up here ONLY if codec is changed.
- memcpy(audio_frame->data_,
- audio_buffer_.get(),
- samples_per_channel * num_channels * sizeof(int16_t));
- }
-
- // Swap buffers, so that the current audio is stored in |last_audio_buffer_|
- // for next time.
- audio_buffer_.swap(last_audio_buffer_);
-
- audio_frame->num_channels_ = num_channels;
- audio_frame->samples_per_channel_ = samples_per_channel;
- audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
-
- // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
- audio_frame->vad_activity_ = previous_audio_activity_;
- SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
- previous_audio_activity_ = audio_frame->vad_activity_;
- call_stats_.DecodedByNetEq(audio_frame->speech_type_);
-
- // Computes the RTP timestamp of the first sample in |audio_frame| from
- // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
- // |audio_frame|.
- uint32_t playout_timestamp = 0;
- if (GetPlayoutTimestamp(&playout_timestamp)) {
- audio_frame->timestamp_ = playout_timestamp -
- static_cast<uint32_t>(audio_frame->samples_per_channel_);
- } else {
- // Remain 0 until we have a valid |playout_timestamp|.
- audio_frame->timestamp_ = 0;
- }
-
- return 0;
-}
-
-int32_t AcmReceiver::AddCodec(int acm_codec_id,
- uint8_t payload_type,
- int channels,
- int sample_rate_hz,
- AudioDecoder* audio_decoder) {
- const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
- if (acm_codec_id == -1)
- return NetEqDecoder::kDecoderArbitrary; // External decoder.
- const rtc::Optional<RentACodec::CodecId> cid =
- RentACodec::CodecIdFromIndex(acm_codec_id);
- RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
- const rtc::Optional<NetEqDecoder> ned =
- RentACodec::NetEqDecoderFromCodecId(*cid, channels);
- RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
- return *ned;
- }();
-
- CriticalSectionScoped lock(crit_sect_.get());
-
- // The corresponding NetEq decoder ID.
- // If this codec has been registered before.
- auto it = decoders_.find(payload_type);
- if (it != decoders_.end()) {
- const Decoder& decoder = it->second;
- if (acm_codec_id != -1 && decoder.acm_codec_id == acm_codec_id &&
- decoder.channels == channels &&
- decoder.sample_rate_hz == sample_rate_hz) {
- // Re-registering the same codec. Do nothing and return.
- return 0;
- }
-
- // Changing codec. First unregister the old codec, then register the new
- // one.
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
- LOG(LERROR) << "Cannot remove payload " << static_cast<int>(payload_type);
- return -1;
- }
-
- decoders_.erase(it);
- }
-
- int ret_val;
- if (!audio_decoder) {
- ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
- } else {
- ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
- payload_type, sample_rate_hz);
- }
- if (ret_val != NetEq::kOK) {
- LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
- << static_cast<int>(payload_type)
- << " channels: " << channels;
- return -1;
- }
-
- Decoder decoder;
- decoder.acm_codec_id = acm_codec_id;
- decoder.payload_type = payload_type;
- decoder.channels = channels;
- decoder.sample_rate_hz = sample_rate_hz;
- decoders_[payload_type] = decoder;
- return 0;
-}
-
-void AcmReceiver::EnableVad() {
- neteq_->EnableVad();
- CriticalSectionScoped lock(crit_sect_.get());
- vad_enabled_ = true;
-}
-
-void AcmReceiver::DisableVad() {
- neteq_->DisableVad();
- CriticalSectionScoped lock(crit_sect_.get());
- vad_enabled_ = false;
-}
-
-void AcmReceiver::FlushBuffers() {
- neteq_->FlushBuffers();
-}
-
-// If failed in removing one of the codecs, this method continues to remove as
-// many as it can.
-int AcmReceiver::RemoveAllCodecs() {
- int ret_val = 0;
- CriticalSectionScoped lock(crit_sect_.get());
- for (auto it = decoders_.begin(); it != decoders_.end(); ) {
- auto cur = it;
- ++it; // it will be valid even if we erase cur
- if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
- decoders_.erase(cur);
- } else {
- LOG_F(LS_ERROR) << "Cannot remove payload "
- << static_cast<int>(cur->second.payload_type);
- ret_val = -1;
- }
- }
-
- // No codec is registered, invalidate last audio decoder.
- last_audio_decoder_ = nullptr;
- last_packet_sample_rate_hz_ = rtc::Optional<int>();
- return ret_val;
-}
-
-int AcmReceiver::RemoveCodec(uint8_t payload_type) {
- CriticalSectionScoped lock(crit_sect_.get());
- auto it = decoders_.find(payload_type);
- if (it == decoders_.end()) { // Such a payload-type is not registered.
- return 0;
- }
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
- LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
- return -1;
- }
- if (last_audio_decoder_ == &it->second) {
- last_audio_decoder_ = nullptr;
- last_packet_sample_rate_hz_ = rtc::Optional<int>();
- }
- decoders_.erase(it);
- return 0;
-}
-
-void AcmReceiver::set_id(int id) {
- CriticalSectionScoped lock(crit_sect_.get());
- id_ = id;
-}
-
-bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
- return neteq_->GetPlayoutTimestamp(timestamp);
-}
-
-int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
- CriticalSectionScoped lock(crit_sect_.get());
- if (!last_audio_decoder_) {
- return -1;
- }
- *codec = *RentACodec::CodecInstById(
- *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
- codec->pltype = last_audio_decoder_->payload_type;
- codec->channels = last_audio_decoder_->channels;
- codec->plfreq = last_audio_decoder_->sample_rate_hz;
- return 0;
-}
-
-void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
- NetEqNetworkStatistics neteq_stat;
- // NetEq function always returns zero, so we don't check the return value.
- neteq_->NetworkStatistics(&neteq_stat);
-
- 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;
- acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
- acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
- 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->clockDriftPPM = neteq_stat.clockdrift_ppm;
- acm_stat->addedSamples = neteq_stat.added_zero_samples;
- acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
- acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
- acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
- acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
-}
-
-int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
- CodecInst* codec) const {
- CriticalSectionScoped lock(crit_sect_.get());
- auto it = decoders_.find(payload_type);
- if (it == decoders_.end()) {
- LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
- << static_cast<int>(payload_type);
- return -1;
- }
- const Decoder& decoder = it->second;
- *codec = *RentACodec::CodecInstById(
- *RentACodec::CodecIdFromIndex(decoder.acm_codec_id));
- codec->pltype = decoder.payload_type;
- codec->channels = decoder.channels;
- codec->plfreq = decoder.sample_rate_hz;
- return 0;
-}
-
-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?
-}
-
-const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
- const RTPHeader& rtp_header,
- uint8_t payload_type) const {
- auto it = decoders_.find(rtp_header.payloadType);
- const auto red_index =
- RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED);
- if (red_index && // This ensures that RED is defined in WebRTC.
- it != decoders_.end() && it->second.acm_codec_id == *red_index) {
- // This is a RED packet, get the payload of the audio codec.
- it = decoders_.find(payload_type & 0x7F);
- }
-
- // Check if the payload is registered.
- return it != decoders_.end() ? &it->second : nullptr;
-}
-
-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 {
- CriticalSectionScoped lock(crit_sect_.get());
- *stats = call_stats_.GetDecodingStatistics();
-}
-
-} // namespace acm2
-
-} // namespace webrtc

Powered by Google App Engine
This is Rietveld 408576698