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webrtc/media/engine/webrtcvoiceengine.cc
Alessio Bazzica cc22f51988 Removing the intelligibility enhancer. 
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).

Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
2018-08-30 21:29:57 +00:00

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/*
* Copyright (c) 2004 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.
*/
#ifdef HAVE_WEBRTC_VOICE
#include "media/engine/webrtcvoiceengine.h"
#include <algorithm>
#include <cstdio>
#include <functional>
#include <string>
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
#include "media/base/audiosource.h"
#include "media/base/mediaconstants.h"
#include "media/base/streamparams.h"
#include "media/engine/adm_helpers.h"
#include "media/engine/apm_helpers.h"
#include "media/engine/payload_type_mapper.h"
#include "media/engine/webrtcmediaengine.h"
#include "modules/audio_device/audio_device_impl.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/byteorder.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace cricket {
namespace {
constexpr size_t kMaxUnsignaledRecvStreams = 4;
constexpr int kNackRtpHistoryMs = 5000;
// For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000.
const int kOpusMinBitrateBps = 6000;
const int kOpusBitrateFbBps = 32000;
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
const int kMaxTelephoneEventCode = 255;
const int kMinPayloadType = 0;
const int kMaxPayloadType = 127;
class ProxySink : public webrtc::AudioSinkInterface {
public:
explicit ProxySink(AudioSinkInterface* sink) : sink_(sink) {
RTC_DCHECK(sink);
}
void OnData(const Data& audio) override { sink_->OnData(audio); }
private:
webrtc::AudioSinkInterface* sink_;
};
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
RTC_LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
return false;
}
if (sp.ssrcs.size() > 1) {
RTC_LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
<< sp.ToString();
return false;
}
return true;
}
// Dumps an AudioCodec in RFC 2327-ish format.
std::string ToString(const AudioCodec& codec) {
std::stringstream ss;
ss << codec.name << "/" << codec.clockrate << "/" << codec.channels;
if (!codec.params.empty()) {
ss << " {";
for (const auto& param : codec.params) {
ss << " " << param.first << "=" << param.second;
}
ss << " }";
}
ss << " (" << codec.id << ")";
return ss.str();
}
bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec) {
for (const AudioCodec& c : codecs) {
if (c.Matches(codec)) {
if (found_codec != NULL) {
*found_codec = c;
}
return true;
}
}
return false;
}
bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
if (codecs.empty()) {
return true;
}
std::vector<int> payload_types;
for (const AudioCodec& codec : codecs) {
payload_types.push_back(codec.id);
}
std::sort(payload_types.begin(), payload_types.end());
auto it = std::unique(payload_types.begin(), payload_types.end());
return it == payload_types.end();
}
absl::optional<std::string> GetAudioNetworkAdaptorConfig(
const AudioOptions& options) {
if (options.audio_network_adaptor && *options.audio_network_adaptor &&
options.audio_network_adaptor_config) {
// Turn on audio network adaptor only when |options_.audio_network_adaptor|
// equals true and |options_.audio_network_adaptor_config| has a value.
return options.audio_network_adaptor_config;
}
return absl::nullopt;
}
// |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
// |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
absl::optional<int> rtp_max_bitrate_bps,
const webrtc::AudioCodecSpec& spec) {
// If application-configured bitrate is set, take minimum of that and SDP
// bitrate.
const int bps =
rtp_max_bitrate_bps
? webrtc::MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
: max_send_bitrate_bps;
if (bps <= 0) {
return spec.info.default_bitrate_bps;
}
if (bps < spec.info.min_bitrate_bps) {
// If codec is not multi-rate and |bps| is less than the fixed bitrate then
// fail. If codec is not multi-rate and |bps| exceeds or equal the fixed
// bitrate then ignore.
RTC_LOG(LS_ERROR) << "Failed to set codec " << spec.format.name
<< " to bitrate " << bps << " bps"
<< ", requires at least " << spec.info.min_bitrate_bps
<< " bps.";
return absl::nullopt;
}
if (spec.info.HasFixedBitrate()) {
return spec.info.default_bitrate_bps;
} else {
// If codec is multi-rate then just set the bitrate.
return std::min(bps, spec.info.max_bitrate_bps);
}
}
} // namespace
WebRtcVoiceEngine::WebRtcVoiceEngine(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
: adm_(adm),
encoder_factory_(encoder_factory),
decoder_factory_(decoder_factory),
audio_mixer_(audio_mixer),
apm_(audio_processing) {
// This may be called from any thread, so detach thread checkers.
worker_thread_checker_.DetachFromThread();
signal_thread_checker_.DetachFromThread();
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
RTC_DCHECK(decoder_factory);
RTC_DCHECK(encoder_factory);
RTC_DCHECK(audio_processing);
// The rest of our initialization will happen in Init.
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
if (initialized_) {
StopAecDump();
// Stop AudioDevice.
adm()->StopPlayout();
adm()->StopRecording();
adm()->RegisterAudioCallback(nullptr);
adm()->Terminate();
}
}
void WebRtcVoiceEngine::Init() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
// TaskQueue expects to be created/destroyed on the same thread.
low_priority_worker_queue_.reset(
new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW));
// Load our audio codec lists.
RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:";
send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
for (const AudioCodec& codec : send_codecs_) {
RTC_LOG(LS_INFO) << ToString(codec);
}
RTC_LOG(LS_INFO) << "Supported recv codecs in order of preference:";
recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
for (const AudioCodec& codec : recv_codecs_) {
RTC_LOG(LS_INFO) << ToString(codec);
}
#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
// No ADM supplied? Create a default one.
if (!adm_) {
adm_ = webrtc::AudioDeviceModule::Create(
webrtc::AudioDeviceModule::kPlatformDefaultAudio);
}
#endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
RTC_CHECK(adm());
webrtc::adm_helpers::Init(adm());
webrtc::apm_helpers::Init(apm());
// Set up AudioState.
{
webrtc::AudioState::Config config;
if (audio_mixer_) {
config.audio_mixer = audio_mixer_;
} else {
config.audio_mixer = webrtc::AudioMixerImpl::Create();
}
config.audio_processing = apm_;
config.audio_device_module = adm_;
audio_state_ = webrtc::AudioState::Create(config);
}
// Connect the ADM to our audio path.
adm()->RegisterAudioCallback(audio_state()->audio_transport());
// Save the default AGC configuration settings. This must happen before
// calling ApplyOptions or the default will be overwritten.
default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm());
// Set default engine options.
{
AudioOptions options;
options.echo_cancellation = true;
options.auto_gain_control = true;
options.noise_suppression = true;
options.highpass_filter = true;
options.stereo_swapping = false;
options.audio_jitter_buffer_max_packets = 50;
options.audio_jitter_buffer_fast_accelerate = false;
options.typing_detection = true;
options.experimental_agc = false;
options.extended_filter_aec = false;
options.delay_agnostic_aec = false;
options.experimental_ns = false;
options.residual_echo_detector = true;
bool error = ApplyOptions(options);
RTC_DCHECK(error);
}
initialized_ = true;
}
rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState()
const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return audio_state_;
}
VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return new WebRtcVoiceMediaChannel(this, config, options, call);
}
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: "
<< options_in.ToString();
AudioOptions options = options_in; // The options are modified below.
// Set and adjust echo canceller options.
// kEcConference is AEC with high suppression.
webrtc::EcModes ec_mode = webrtc::kEcConference;
if (options.aecm_generate_comfort_noise) {
RTC_LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
<< *options.aecm_generate_comfort_noise
<< " (default is false).";
}
#if defined(WEBRTC_IOS)
if (options.ios_force_software_aec_HACK &&
*options.ios_force_software_aec_HACK) {
// EC may be forced on for a device known to have non-functioning platform
// AEC.
options.echo_cancellation = true;
options.extended_filter_aec = true;
RTC_LOG(LS_WARNING)
<< "Force software AEC on iOS. May conflict with platform AEC.";
} else {
// On iOS, VPIO provides built-in EC.
options.echo_cancellation = false;
options.extended_filter_aec = false;
RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead.";
}
#elif defined(WEBRTC_ANDROID)
ec_mode = webrtc::kEcAecm;
options.extended_filter_aec = false;
#endif
// Delay Agnostic AEC automatically turns on EC if not set except on iOS
// where the feature is not supported.
bool use_delay_agnostic_aec = false;
#if !defined(WEBRTC_IOS)
if (options.delay_agnostic_aec) {
use_delay_agnostic_aec = *options.delay_agnostic_aec;
if (use_delay_agnostic_aec) {
options.echo_cancellation = true;
options.extended_filter_aec = true;
ec_mode = webrtc::kEcConference;
}
}
#endif
// Set and adjust noise suppressor options.
#if defined(WEBRTC_IOS)
// On iOS, VPIO provides built-in NS.
options.noise_suppression = false;
options.typing_detection = false;
options.experimental_ns = false;
RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
options.typing_detection = false;
options.experimental_ns = false;
#endif
// Set and adjust gain control options.
#if defined(WEBRTC_IOS)
// On iOS, VPIO provides built-in AGC.
options.auto_gain_control = false;
options.experimental_agc = false;
RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
options.experimental_agc = false;
#endif
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
// Turn off the gain control if specified by the field trial.
// The purpose of the field trial is to reduce the amount of resampling
// performed inside the audio processing module on mobile platforms by
// whenever possible turning off the fixed AGC mode and the high-pass filter.
// (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181).
if (webrtc::field_trial::IsEnabled(
"WebRTC-Audio-MinimizeResamplingOnMobile")) {
options.auto_gain_control = false;
RTC_LOG(LS_INFO) << "Disable AGC according to field trial.";
if (!(options.noise_suppression.value_or(false) ||
options.echo_cancellation.value_or(false))) {
// If possible, turn off the high-pass filter.
RTC_LOG(LS_INFO)
<< "Disable high-pass filter in response to field trial.";
options.highpass_filter = false;
}
}
#endif
if (options.echo_cancellation) {
// Check if platform supports built-in EC. Currently only supported on
// Android and in combination with Java based audio layer.
// TODO(henrika): investigate possibility to support built-in EC also
// in combination with Open SL ES audio.
const bool built_in_aec = adm()->BuiltInAECIsAvailable();
if (built_in_aec) {
// Built-in EC exists on this device and use_delay_agnostic_aec is not
// overriding it. Enable/Disable it according to the echo_cancellation
// audio option.
const bool enable_built_in_aec =
*options.echo_cancellation && !use_delay_agnostic_aec;
if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
options.echo_cancellation = false;
RTC_LOG(LS_INFO)
<< "Disabling EC since built-in EC will be used instead";
}
}
webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation,
ec_mode);
#if !defined(WEBRTC_ANDROID)
webrtc::apm_helpers::SetEcMetricsStatus(apm(), *options.echo_cancellation);
#endif
if (ec_mode == webrtc::kEcAecm) {
bool cn = options.aecm_generate_comfort_noise.value_or(false);
webrtc::apm_helpers::SetAecmMode(apm(), cn);
}
}
if (options.auto_gain_control) {
bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();
if (built_in_agc_avaliable) {
if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
*options.auto_gain_control) {
// Disable internal software AGC if built-in AGC is enabled,
// i.e., replace the software AGC with the built-in AGC.
options.auto_gain_control = false;
RTC_LOG(LS_INFO)
<< "Disabling AGC since built-in AGC will be used instead";
}
}
webrtc::apm_helpers::SetAgcStatus(apm(), *options.auto_gain_control);
}
if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
options.tx_agc_limiter) {
// Override default_agc_config_. Generally, an unset option means "leave
// the VoE bits alone" in this function, so we want whatever is set to be
// stored as the new "default". If we didn't, then setting e.g.
// tx_agc_target_dbov would reset digital compression gain and limiter
// settings.
default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or(
default_agc_config_.targetLeveldBOv);
default_agc_config_.digitalCompressionGaindB =
options.tx_agc_digital_compression_gain.value_or(
default_agc_config_.digitalCompressionGaindB);
default_agc_config_.limiterEnable =
options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);
webrtc::apm_helpers::SetAgcConfig(apm(), default_agc_config_);
}
if (options.noise_suppression) {
if (adm()->BuiltInNSIsAvailable()) {
bool builtin_ns = *options.noise_suppression;
if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
// Disable internal software NS if built-in NS is enabled,
// i.e., replace the software NS with the built-in NS.
options.noise_suppression = false;
RTC_LOG(LS_INFO)
<< "Disabling NS since built-in NS will be used instead";
}
}
webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression);
}
if (options.stereo_swapping) {
RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
audio_state()->SetStereoChannelSwapping(*options.stereo_swapping);
}
if (options.audio_jitter_buffer_max_packets) {
RTC_LOG(LS_INFO) << "NetEq capacity is "
<< *options.audio_jitter_buffer_max_packets;
audio_jitter_buffer_max_packets_ =
std::max(20, *options.audio_jitter_buffer_max_packets);
}
if (options.audio_jitter_buffer_fast_accelerate) {
RTC_LOG(LS_INFO) << "NetEq fast mode? "
<< *options.audio_jitter_buffer_fast_accelerate;
audio_jitter_buffer_fast_accelerate_ =
*options.audio_jitter_buffer_fast_accelerate;
}
if (options.typing_detection) {
RTC_LOG(LS_INFO) << "Typing detection is enabled? "
<< *options.typing_detection;
webrtc::apm_helpers::SetTypingDetectionStatus(apm(),
*options.typing_detection);
}
webrtc::Config config;
if (options.delay_agnostic_aec)
delay_agnostic_aec_ = options.delay_agnostic_aec;
if (delay_agnostic_aec_) {
RTC_LOG(LS_INFO) << "Delay agnostic aec is enabled? "
<< *delay_agnostic_aec_;
config.Set<webrtc::DelayAgnostic>(
new webrtc::DelayAgnostic(*delay_agnostic_aec_));
}
if (options.extended_filter_aec) {
extended_filter_aec_ = options.extended_filter_aec;
}
if (extended_filter_aec_) {
RTC_LOG(LS_INFO) << "Extended filter aec is enabled? "
<< *extended_filter_aec_;
config.Set<webrtc::ExtendedFilter>(
new webrtc::ExtendedFilter(*extended_filter_aec_));
}
if (options.experimental_ns) {
experimental_ns_ = options.experimental_ns;
}
if (experimental_ns_) {
RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
config.Set<webrtc::ExperimentalNs>(
new webrtc::ExperimentalNs(*experimental_ns_));
}
webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();
if (options.highpass_filter) {
apm_config.high_pass_filter.enabled = *options.highpass_filter;
}
if (options.residual_echo_detector) {
apm_config.residual_echo_detector.enabled = *options.residual_echo_detector;
}
apm()->SetExtraOptions(config);
apm()->ApplyConfig(apm_config);
return true;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return send_codecs_;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return recv_codecs_;
}
RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAudioLevelDefaultId));
if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") &&
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
capabilities.header_extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
}
// TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
// demuxing is completed.
// capabilities.header_extensions.push_back(webrtc::RtpExtension(
// webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
return capabilities;
}
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(channel);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = std::find(channels_.begin(), channels_.end(), channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto aec_dump = webrtc::AecDumpFactory::Create(
file, max_size_bytes, low_priority_worker_queue_.get());
if (!aec_dump) {
return false;
}
apm()->AttachAecDump(std::move(aec_dump));
return true;
}
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto aec_dump = webrtc::AecDumpFactory::Create(
filename, -1, low_priority_worker_queue_.get());
if (aec_dump) {
apm()->AttachAecDump(std::move(aec_dump));
}
}
void WebRtcVoiceEngine::StopAecDump() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
apm()->DetachAecDump();
}
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(adm_);
return adm_.get();
}
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(apm_);
return apm_.get();
}
webrtc::AudioState* WebRtcVoiceEngine::audio_state() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(audio_state_);
return audio_state_.get();
}
AudioCodecs WebRtcVoiceEngine::CollectCodecs(
const std::vector<webrtc::AudioCodecSpec>& specs) const {
PayloadTypeMapper mapper;
AudioCodecs out;
// Only generate CN payload types for these clockrates:
std::map<int, bool, std::greater<int>> generate_cn = {
{8000, false}, {16000, false}, {32000, false}};
// Only generate telephone-event payload types for these clockrates:
std::map<int, bool, std::greater<int>> generate_dtmf = {
{8000, false}, {16000, false}, {32000, false}, {48000, false}};
auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
AudioCodecs* out) {
absl::optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
if (opt_codec) {
if (out) {
out->push_back(*opt_codec);
}
} else {
RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: "
<< rtc::ToString(format);
}
return opt_codec;
};
for (const auto& spec : specs) {
// We need to do some extra stuff before adding the main codecs to out.
absl::optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
if (opt_codec) {
AudioCodec& codec = *opt_codec;
if (spec.info.supports_network_adaption) {
codec.AddFeedbackParam(
FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
}
if (spec.info.allow_comfort_noise) {
// Generate a CN entry if the decoder allows it and we support the
// clockrate.
auto cn = generate_cn.find(spec.format.clockrate_hz);
if (cn != generate_cn.end()) {
cn->second = true;
}
}
// Generate a telephone-event entry if we support the clockrate.
auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
if (dtmf != generate_dtmf.end()) {
dtmf->second = true;
}
out.push_back(codec);
}
}
// Add CN codecs after "proper" audio codecs.
for (const auto& cn : generate_cn) {
if (cn.second) {
map_format({kCnCodecName, cn.first, 1}, &out);
}
}
// Add telephone-event codecs last.
for (const auto& dtmf : generate_dtmf) {
if (dtmf.second) {
map_format({kDtmfCodecName, dtmf.first, 1}, &out);
}
}
return out;
}
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioSource::Sink {
public:
WebRtcAudioSendStream(
uint32_t ssrc,
const std::string& mid,
const std::string& c_name,
const std::string track_id,
const absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
send_codec_spec,
const std::vector<webrtc::RtpExtension>& extensions,
int max_send_bitrate_bps,
const absl::optional<std::string>& audio_network_adaptor_config,
webrtc::Call* call,
webrtc::Transport* send_transport,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const absl::optional<webrtc::AudioCodecPairId> codec_pair_id)
: call_(call),
config_(send_transport),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
max_send_bitrate_bps_(max_send_bitrate_bps),
rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
RTC_DCHECK(call);
RTC_DCHECK(encoder_factory);
config_.rtp.ssrc = ssrc;
config_.rtp.mid = mid;
config_.rtp.c_name = c_name;
config_.rtp.extensions = extensions;
config_.audio_network_adaptor_config = audio_network_adaptor_config;
config_.encoder_factory = encoder_factory;
config_.codec_pair_id = codec_pair_id;
config_.track_id = track_id;
rtp_parameters_.encodings[0].ssrc = ssrc;
rtp_parameters_.rtcp.cname = c_name;
rtp_parameters_.header_extensions = extensions;
if (send_codec_spec) {
UpdateSendCodecSpec(*send_codec_spec);
}
stream_ = call_->CreateAudioSendStream(config_);
}
~WebRtcAudioSendStream() override {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
ClearSource();
call_->DestroyAudioSendStream(stream_);
}
void SetSendCodecSpec(
const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
UpdateSendCodecSpec(send_codec_spec);
ReconfigureAudioSendStream();
}
void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.extensions = extensions;
rtp_parameters_.header_extensions = extensions;
ReconfigureAudioSendStream();
}
void SetMid(const std::string& mid) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (config_.rtp.mid == mid) {
return;
}
config_.rtp.mid = mid;
ReconfigureAudioSendStream();
}
void SetAudioNetworkAdaptorConfig(
const absl::optional<std::string>& audio_network_adaptor_config) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (config_.audio_network_adaptor_config == audio_network_adaptor_config) {
return;
}
config_.audio_network_adaptor_config = audio_network_adaptor_config;
UpdateAllowedBitrateRange();
ReconfigureAudioSendStream();
}
bool SetMaxSendBitrate(int bps) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(config_.send_codec_spec);
RTC_DCHECK(audio_codec_spec_);
auto send_rate = ComputeSendBitrate(
bps, rtp_parameters_.encodings[0].max_bitrate_bps, *audio_codec_spec_);
if (!send_rate) {
return false;
}
max_send_bitrate_bps_ = bps;
if (send_rate != config_.send_codec_spec->target_bitrate_bps) {
config_.send_codec_spec->target_bitrate_bps = send_rate;
ReconfigureAudioSendStream();
}
return true;
}
bool SendTelephoneEvent(int payload_type,
int payload_freq,
int event,
int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->SendTelephoneEvent(payload_type, payload_freq, event,
duration_ms);
}
void SetSend(bool send) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
send_ = send;
UpdateSendState();
}
void SetMuted(bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
stream_->SetMuted(muted);
muted_ = muted;
}
bool muted() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return muted_;
}
webrtc::AudioSendStream::Stats GetStats(bool has_remote_tracks) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetStats(has_remote_tracks);
}
// Starts the sending by setting ourselves as a sink to the AudioSource to
// get data callbacks.
// This method is called on the libjingle worker thread.
// TODO(xians): Make sure Start() is called only once.
void SetSource(AudioSource* source) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(source);
if (source_) {
RTC_DCHECK(source_ == source);
return;
}
source->SetSink(this);
source_ = source;
UpdateSendState();
}
// Stops sending by setting the sink of the AudioSource to nullptr. No data
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void ClearSource() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (source_) {
source_->SetSink(nullptr);
source_ = nullptr;
}
UpdateSendState();
}
// AudioSource::Sink implementation.
// This method is called on the audio thread.
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {
RTC_DCHECK_EQ(16, bits_per_sample);
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK(stream_);
std::unique_ptr<webrtc::AudioFrame> audio_frame(new webrtc::AudioFrame());
audio_frame->UpdateFrame(
audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),
number_of_frames, sample_rate, audio_frame->speech_type_,
audio_frame->vad_activity_, number_of_channels);
stream_->SendAudioData(std::move(audio_frame));
}
// Callback from the |source_| when it is going away. In case Start() has
// never been called, this callback won't be triggered.
void OnClose() override {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set |source_| to nullptr to make sure no more callback will get into
// the source.
source_ = nullptr;
UpdateSendState();
}
const webrtc::RtpParameters& rtp_parameters() const {
return rtp_parameters_;
}
webrtc::RTCError ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
using webrtc::RTCErrorType;
if (rtp_parameters.encodings.size() != rtp_parameters_.encodings.size()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with different encoding count");
}
if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with modified RTCP parameters");
}
if (rtp_parameters.header_extensions != rtp_parameters_.header_extensions) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with modified header extensions");
}
if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Attempted to set RtpParameters with modified SSRC");
}
if (rtp_parameters.encodings[0].bitrate_priority <= 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
"Attempted to set RtpParameters bitrate_priority to "
"an invalid number.");
}
return webrtc::RTCError::OK();
}
webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters) {
webrtc::RTCError error = ValidateRtpParameters(parameters);
if (!error.ok()) {
return error;
}
absl::optional<int> send_rate;
if (audio_codec_spec_) {
send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
parameters.encodings[0].max_bitrate_bps,
*audio_codec_spec_);
if (!send_rate) {
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
}
const absl::optional<int> old_rtp_max_bitrate =
rtp_parameters_.encodings[0].max_bitrate_bps;
double old_priority = rtp_parameters_.encodings[0].bitrate_priority;
rtp_parameters_ = parameters;
config_.bitrate_priority = rtp_parameters_.encodings[0].bitrate_priority;
bool reconfigure_send_stream =
(rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) ||
(rtp_parameters_.encodings[0].bitrate_priority != old_priority);
if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) {
// Update the bitrate range.
if (send_rate) {
config_.send_codec_spec->target_bitrate_bps = send_rate;
}
UpdateAllowedBitrateRange();
}
if (reconfigure_send_stream) {
ReconfigureAudioSendStream();
}
rtp_parameters_.rtcp.cname = config_.rtp.c_name;
rtp_parameters_.rtcp.reduced_size = false;
// parameters.encodings[0].active could have changed.
UpdateSendState();
return webrtc::RTCError::OK();
}
private:
void UpdateSendState() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
stream_->Start();
} else { // !send || source_ = nullptr
stream_->Stop();
}
}
void UpdateAllowedBitrateRange() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const bool is_opus =
config_.send_codec_spec &&
!STR_CASE_CMP(config_.send_codec_spec->format.name.c_str(),
kOpusCodecName);
if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
config_.min_bitrate_bps = kOpusMinBitrateBps;
// This means that when RtpParameters is reset, we may change the
// encoder's bit rate immediately (through ReconfigureAudioSendStream()),
// meanwhile change the cap to the output of BWE.
config_.max_bitrate_bps =
rtp_parameters_.encodings[0].max_bitrate_bps
? *rtp_parameters_.encodings[0].max_bitrate_bps
: kOpusBitrateFbBps;
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
if (send_side_bwe_with_overhead_) {
const int max_packet_size_ms =
WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
int min_overhead_bps =
kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
// We assume that |config_.max_bitrate_bps| before the next line is
// a hard limit on the payload bitrate, so we add min_overhead_bps to
// it to ensure that, when overhead is deducted, the payload rate
// never goes beyond the limit.
// Note: this also means that if a higher overhead is forced, we
// cannot reach the limit.
// TODO(minyue): Reconsider this when the signaling to BWE is done
// through a dedicated API.
config_.max_bitrate_bps += min_overhead_bps;
// In contrast to max_bitrate_bps, we let min_bitrate_bps always be
// reachable.
config_.min_bitrate_bps += min_overhead_bps;
}
}
}
void UpdateSendCodecSpec(
const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.nack.rtp_history_ms =
send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
config_.send_codec_spec = send_codec_spec;
auto info =
config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format);
RTC_DCHECK(info);
// If a specific target bitrate has been set for the stream, use that as
// the new default bitrate when computing send bitrate.
if (send_codec_spec.target_bitrate_bps) {
info->default_bitrate_bps = std::max(
info->min_bitrate_bps,
std::min(info->max_bitrate_bps, *send_codec_spec.target_bitrate_bps));
}
audio_codec_spec_.emplace(
webrtc::AudioCodecSpec{send_codec_spec.format, *info});
config_.send_codec_spec->target_bitrate_bps = ComputeSendBitrate(
max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps,
*audio_codec_spec_);
UpdateAllowedBitrateRange();
}
void ReconfigureAudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
stream_->Reconfigure(config_);
}
rtc::ThreadChecker worker_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
webrtc::Call* call_ = nullptr;
webrtc::AudioSendStream::Config config_;
const bool send_side_bwe_with_overhead_;
// The stream is owned by WebRtcAudioSendStream and may be reallocated if
// configuration changes.
webrtc::AudioSendStream* stream_ = nullptr;
// Raw pointer to AudioSource owned by LocalAudioTrackHandler.
// PeerConnection will make sure invalidating the pointer before the object
// goes away.
AudioSource* source_ = nullptr;
bool send_ = false;
bool muted_ = false;
int max_send_bitrate_bps_;
webrtc::RtpParameters rtp_parameters_;
absl::optional<webrtc::AudioCodecSpec> audio_codec_spec_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};
class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
public:
WebRtcAudioReceiveStream(
uint32_t remote_ssrc,
uint32_t local_ssrc,
bool use_transport_cc,
bool use_nack,
const std::vector<std::string>& stream_ids,
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call,
webrtc::Transport* rtcp_send_transport,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_accelerate)
: call_(call), config_() {
RTC_DCHECK(call);
config_.rtp.remote_ssrc = remote_ssrc;
config_.rtp.local_ssrc = local_ssrc;
config_.rtp.transport_cc = use_transport_cc;
config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
config_.rtp.extensions = extensions;
config_.rtcp_send_transport = rtcp_send_transport;
config_.jitter_buffer_max_packets = jitter_buffer_max_packets;
config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
if (!stream_ids.empty()) {
config_.sync_group = stream_ids[0];
}
config_.decoder_factory = decoder_factory;
config_.decoder_map = decoder_map;
config_.codec_pair_id = codec_pair_id;
RecreateAudioReceiveStream();
}
~WebRtcAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
call_->DestroyAudioReceiveStream(stream_);
}
void SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.local_ssrc = local_ssrc;
ReconfigureAudioReceiveStream();
}
void SetUseTransportCcAndRecreateStream(bool use_transport_cc,
bool use_nack) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.transport_cc = use_transport_cc;
config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
ReconfigureAudioReceiveStream();
}
void SetRtpExtensionsAndRecreateStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.extensions = extensions;
RecreateAudioReceiveStream();
}
// Set a new payload type -> decoder map.
void SetDecoderMap(const std::map<int, webrtc::SdpAudioFormat>& decoder_map) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.decoder_map = decoder_map;
ReconfigureAudioReceiveStream();
}
void MaybeRecreateAudioReceiveStream(
const std::vector<std::string>& stream_ids) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
std::string sync_group;
if (!stream_ids.empty()) {
sync_group = stream_ids[0];
}
if (config_.sync_group != sync_group) {
RTC_LOG(LS_INFO) << "Recreating AudioReceiveStream for SSRC="
<< config_.rtp.remote_ssrc
<< " because of sync group change.";
config_.sync_group = sync_group;
RecreateAudioReceiveStream();
}
}
webrtc::AudioReceiveStream::Stats GetStats() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetStats();
}
void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Need to update the stream's sink first; once raw_audio_sink_ is
// reassigned, whatever was in there before is destroyed.
stream_->SetSink(sink.get());
raw_audio_sink_ = std::move(sink);
}
void SetOutputVolume(double volume) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
output_volume_ = volume;
stream_->SetGain(volume);
}
void SetPlayout(bool playout) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
if (playout) {
stream_->Start();
} else {
stream_->Stop();
}
playout_ = playout;
}
std::vector<webrtc::RtpSource> GetSources() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetSources();
}
webrtc::RtpParameters GetRtpParameters() const {
webrtc::RtpParameters rtp_parameters;
rtp_parameters.encodings.emplace_back();
rtp_parameters.encodings[0].ssrc = config_.rtp.remote_ssrc;
rtp_parameters.header_extensions = config_.rtp.extensions;
return rtp_parameters;
}
private:
void RecreateAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioReceiveStream(stream_);
}
stream_ = call_->CreateAudioReceiveStream(config_);
RTC_CHECK(stream_);
stream_->SetGain(output_volume_);
SetPlayout(playout_);
stream_->SetSink(raw_audio_sink_.get());
}
void ReconfigureAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
stream_->Reconfigure(config_);
}
rtc::ThreadChecker worker_thread_checker_;
webrtc::Call* call_ = nullptr;
webrtc::AudioReceiveStream::Config config_;
// The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
// configuration changes.
webrtc::AudioReceiveStream* stream_ = nullptr;
bool playout_ = false;
float output_volume_ = 1.0;
std::unique_ptr<webrtc::AudioSinkInterface> raw_audio_sink_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
webrtc::Call* call)
: VoiceMediaChannel(config), engine_(engine), call_(call) {
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
engine->RegisterChannel(this);
SetOptions(options);
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
// TODO(solenberg): Should be able to delete the streams directly, without
// going through RemoveNnStream(), once stream objects handle
// all (de)configuration.
while (!send_streams_.empty()) {
RemoveSendStream(send_streams_.begin()->first);
}
while (!recv_streams_.empty()) {
RemoveRecvStream(recv_streams_.begin()->first);
}
engine()->UnregisterChannel(this);
}
rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
return kAudioDscpValue;
}
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
<< params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
if (!SetSendCodecs(params.codecs)) {
return false;
}
if (!ValidateRtpExtensions(params.extensions)) {
return false;
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true);
if (send_rtp_extensions_ != filtered_extensions) {
send_rtp_extensions_.swap(filtered_extensions);
for (auto& it : send_streams_) {
it.second->SetRtpExtensions(send_rtp_extensions_);
}
}
if (!params.mid.empty()) {
mid_ = params.mid;
for (auto& it : send_streams_) {
it.second->SetMid(params.mid);
}
}
if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
return false;
}
return SetOptions(params.options);
}
bool WebRtcVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
<< params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
if (!SetRecvCodecs(params.codecs)) {
return false;
}
if (!ValidateRtpExtensions(params.extensions)) {
return false;
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, false);
if (recv_rtp_extensions_ != filtered_extensions) {
recv_rtp_extensions_.swap(filtered_extensions);
for (auto& it : recv_streams_) {
it.second->SetRtpExtensionsAndRecreateStream(recv_rtp_extensions_);
}
}
return true;
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
webrtc::RtpParameters rtp_params = it->second->rtp_parameters();
// Need to add the common list of codecs to the send stream-specific
// RTP parameters.
for (const AudioCodec& codec : send_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
// TODO(deadbeef): Handle setting parameters with a list of codecs in a
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
}
// TODO(minyue): The following legacy actions go into
// |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
// though there are two difference:
// 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls
// |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls
// |SetSendCodecs|. The outcome should be the same.
// 2. AudioSendStream can be recreated.
// Codecs are handled at the WebRtcVoiceMediaChannel level.
webrtc::RtpParameters reduced_params = parameters;
reduced_params.codecs.clear();
return it->second->SetRtpParameters(reduced_params);
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::RtpParameters rtp_params;
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
if (!default_sink_) {
RTC_LOG(LS_WARNING)
<< "Attempting to get RTP parameters for the default, "
"unsignaled audio receive stream, but not yet "
"configured to receive such a stream.";
return rtp_params;
}
rtp_params.encodings.emplace_back();
} else {
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING)
<< "Attempting to get RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
rtp_params = it->second->GetRtpParameters();
}
for (const AudioCodec& codec : recv_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
if (!default_sink_) {
RTC_LOG(LS_WARNING)
<< "Attempting to set RTP parameters for the default, "
"unsignaled audio receive stream, but not yet "
"configured to receive such a stream.";
return false;
}
} else {
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING)
<< "Attempting to set RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
}
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "Setting voice channel options: " << options.ToString();
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
options_.SetAll(options);
if (!engine()->ApplyOptions(options_)) {
RTC_LOG(LS_WARNING)
<< "Failed to apply engine options during channel SetOptions.";
return false;
}
absl::optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
for (auto& it : send_streams_) {
it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config);
}
RTC_LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
}
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set the payload types to be used for incoming media.
RTC_LOG(LS_INFO) << "Setting receive voice codecs.";
if (!VerifyUniquePayloadTypes(codecs)) {
RTC_LOG(LS_ERROR) << "Codec payload types overlap.";
return false;
}
// Create a payload type -> SdpAudioFormat map with all the decoders. Fail
// unless the factory claims to support all decoders.
std::map<int, webrtc::SdpAudioFormat> decoder_map;
for (const AudioCodec& codec : codecs) {
// Log a warning if a codec's payload type is changing. This used to be
// treated as an error. It's abnormal, but not really illegal.
AudioCodec old_codec;
if (FindCodec(recv_codecs_, codec, &old_codec) &&
old_codec.id != codec.id) {
RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type ("
<< codec.id << ", was already mapped to "
<< old_codec.id << ")";
}
auto format = AudioCodecToSdpAudioFormat(codec);
if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") &&
!engine()->decoder_factory_->IsSupportedDecoder(format)) {
RTC_LOG(LS_ERROR) << "Unsupported codec: " << rtc::ToString(format);
return false;
}
// We allow adding new codecs but don't allow changing the payload type of
// codecs that are already configured since we might already be receiving
// packets with that payload type. See RFC3264, Section 8.3.2.
// TODO(deadbeef): Also need to check for clashes with previously mapped
// payload types, and not just currently mapped ones. For example, this
// should be illegal:
// 1. {100: opus/48000/2, 101: ISAC/16000}
// 2. {100: opus/48000/2}
// 3. {100: opus/48000/2, 101: ISAC/32000}
// Though this check really should happen at a higher level, since this
// conflict could happen between audio and video codecs.
auto existing = decoder_map_.find(codec.id);
if (existing != decoder_map_.end() && !existing->second.Matches(format)) {
RTC_LOG(LS_ERROR) << "Attempting to use payload type " << codec.id
<< " for " << codec.name
<< ", but it is already used for "
<< existing->second.name;
return false;
}
decoder_map.insert({codec.id, std::move(format)});
}
if (decoder_map == decoder_map_) {
// There's nothing new to configure.
return true;
}
if (playout_) {
// Receive codecs can not be changed while playing. So we temporarily
// pause playout.
ChangePlayout(false);
}
decoder_map_ = std::move(decoder_map);
for (auto& kv : recv_streams_) {
kv.second->SetDecoderMap(decoder_map_);
}
recv_codecs_ = codecs;
if (desired_playout_ && !playout_) {
ChangePlayout(desired_playout_);
}
return true;
}
// Utility function called from SetSendParameters() to extract current send
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
dtmf_payload_type_ = absl::nullopt;
dtmf_payload_freq_ = -1;
// Validate supplied codecs list.
for (const AudioCodec& codec : codecs) {
// TODO(solenberg): Validate more aspects of input - that payload types
// don't overlap, remove redundant/unsupported codecs etc -
// the same way it is done for RtpHeaderExtensions.
if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
RTC_LOG(LS_WARNING) << "Codec payload type out of range: "
<< ToString(codec);
return false;
}
}
// Find PT of telephone-event codec with lowest clockrate, as a fallback, in
// case we don't have a DTMF codec with a rate matching the send codec's, or
// if this function returns early.
std::vector<AudioCodec> dtmf_codecs;
for (const AudioCodec& codec : codecs) {
if (IsCodec(codec, kDtmfCodecName)) {
dtmf_codecs.push_back(codec);
if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) {
dtmf_payload_type_ = codec.id;
dtmf_payload_freq_ = codec.clockrate;
}
}
}
// Scan through the list to figure out the codec to use for sending.
absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
send_codec_spec;
webrtc::BitrateConstraints bitrate_config;
absl::optional<webrtc::AudioCodecInfo> voice_codec_info;
for (const AudioCodec& voice_codec : codecs) {
if (!(IsCodec(voice_codec, kCnCodecName) ||
IsCodec(voice_codec, kDtmfCodecName) ||
IsCodec(voice_codec, kRedCodecName))) {
webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate,
voice_codec.channels, voice_codec.params);
voice_codec_info = engine()->encoder_factory_->QueryAudioEncoder(format);
if (!voice_codec_info) {
RTC_LOG(LS_WARNING) << "Unknown codec " << ToString(voice_codec);
continue;
}
send_codec_spec = webrtc::AudioSendStream::Config::SendCodecSpec(
voice_codec.id, format);
if (voice_codec.bitrate > 0) {
send_codec_spec->target_bitrate_bps = voice_codec.bitrate;
}
send_codec_spec->transport_cc_enabled = HasTransportCc(voice_codec);
send_codec_spec->nack_enabled = HasNack(voice_codec);
bitrate_config = GetBitrateConfigForCodec(voice_codec);
break;
}
}
if (!send_codec_spec) {
return false;
}
RTC_DCHECK(voice_codec_info);
if (voice_codec_info->allow_comfort_noise) {
// Loop through the codecs list again to find the CN codec.
// TODO(solenberg): Break out into a separate function?
for (const AudioCodec& cn_codec : codecs) {
if (IsCodec(cn_codec, kCnCodecName) &&
cn_codec.clockrate == send_codec_spec->format.clockrate_hz) {
switch (cn_codec.clockrate) {
case 8000:
case 16000:
case 32000:
send_codec_spec->cng_payload_type = cn_codec.id;
break;
default:
RTC_LOG(LS_WARNING)
<< "CN frequency " << cn_codec.clockrate << " not supported.";
break;
}
break;
}
}
// Find the telephone-event PT exactly matching the preferred send codec.
for (const AudioCodec& dtmf_codec : dtmf_codecs) {
if (dtmf_codec.clockrate == send_codec_spec->format.clockrate_hz) {
dtmf_payload_type_ = dtmf_codec.id;
dtmf_payload_freq_ = dtmf_codec.clockrate;
break;
}
}
}
if (send_codec_spec_ != send_codec_spec) {
send_codec_spec_ = std::move(send_codec_spec);
// Apply new settings to all streams.
for (const auto& kv : send_streams_) {
kv.second->SetSendCodecSpec(*send_codec_spec_);
}
} else {
// If the codec isn't changing, set the start bitrate to -1 which means
// "unchanged" so that BWE isn't affected.
bitrate_config.start_bitrate_bps = -1;
}
call_->GetTransportControllerSend()->SetSdpBitrateParameters(bitrate_config);
// Check if the transport cc feedback or NACK status has changed on the
// preferred send codec, and in that case reconfigure all receive streams.
if (recv_transport_cc_enabled_ != send_codec_spec_->transport_cc_enabled ||
recv_nack_enabled_ != send_codec_spec_->nack_enabled) {
RTC_LOG(LS_INFO) << "Recreate all the receive streams because the send "
"codec has changed.";
recv_transport_cc_enabled_ = send_codec_spec_->transport_cc_enabled;
recv_nack_enabled_ = send_codec_spec_->nack_enabled;
for (auto& kv : recv_streams_) {
kv.second->SetUseTransportCcAndRecreateStream(recv_transport_cc_enabled_,
recv_nack_enabled_);
}
}
send_codecs_ = codecs;
return true;
}
void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
desired_playout_ = playout;
return ChangePlayout(desired_playout_);
}
void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (playout_ == playout) {
return;
}
for (const auto& kv : recv_streams_) {
kv.second->SetPlayout(playout);
}
playout_ = playout;
}
void WebRtcVoiceMediaChannel::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
if (send_ == send) {
return;
}
// Apply channel specific options, and initialize the ADM for recording (this
// may take time on some platforms, e.g. Android).
if (send) {
engine()->ApplyOptions(options_);
// InitRecording() may return an error if the ADM is already recording.
if (!engine()->adm()->RecordingIsInitialized() &&
!engine()->adm()->Recording()) {
if (engine()->adm()->InitRecording() != 0) {
RTC_LOG(LS_WARNING) << "Failed to initialize recording";
}
}
}
// Change the settings on each send channel.
for (auto& kv : send_streams_) {
kv.second->SetSend(send);
}
send_ = send;
}
bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// TODO(solenberg): The state change should be fully rolled back if any one of
// these calls fail.
if (!SetLocalSource(ssrc, source)) {
return false;
}
if (!MuteStream(ssrc, !enable)) {
return false;
}
if (enable && options) {
return SetOptions(*options);
}
return true;
}
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(0 != ssrc);
if (send_streams_.find(ssrc) != send_streams_.end()) {
RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
absl::optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
ssrc, mid_, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
engine()->encoder_factory_, codec_pair_id_);
send_streams_.insert(std::make_pair(ssrc, stream));
// At this point the stream's local SSRC has been updated. If it is the first
// send stream, make sure that all the receive streams are updated with the
// same SSRC in order to send receiver reports.
if (send_streams_.size() == 1) {
receiver_reports_ssrc_ = ssrc;
for (const auto& kv : recv_streams_) {
// TODO(solenberg): Allow applications to set the RTCP SSRC of receive
// streams instead, so we can avoid reconfiguring the streams here.
kv.second->SetLocalSsrc(ssrc);
}
}
send_streams_[ssrc]->SetSend(send_);
return true;
}
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
it->second->SetSend(false);
// TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find
// the first active send stream and use that instead, reassociating receive
// streams.
delete it->second;
send_streams_.erase(it);
if (send_streams_.empty()) {
SetSend(false);
}
return true;
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
if (!sp.has_ssrcs()) {
// This is a StreamParam with unsignaled SSRCs. Store it, so it can be used
// later when we know the SSRCs on the first packet arrival.
unsignaled_stream_params_ = sp;
return true;
}
if (!ValidateStreamParams(sp)) {
return false;
}
const uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
RTC_LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
return false;
}
// If this stream was previously received unsignaled, we promote it, possibly
// recreating the AudioReceiveStream, if stream ids have changed.
if (MaybeDeregisterUnsignaledRecvStream(ssrc)) {
recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.stream_ids());
return true;
}
if (recv_streams_.find(ssrc) != recv_streams_.end()) {
RTC_LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
// Create a new channel for receiving audio data.
recv_streams_.insert(std::make_pair(
ssrc, new WebRtcAudioReceiveStream(
ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
call_, this, engine()->decoder_factory_, decoder_map_,
codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
engine()->audio_jitter_buffer_fast_accelerate_)));
recv_streams_[ssrc]->SetPlayout(playout_);
return true;
}
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
if (ssrc == 0) {
// This indicates that we need to remove the unsignaled stream parameters
// that are cached.
unsignaled_stream_params_ = StreamParams();
return true;
}
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
MaybeDeregisterUnsignaledRecvStream(ssrc);
it->second->SetRawAudioSink(nullptr);
delete it->second;
recv_streams_.erase(it);
return true;
}
bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
AudioSource* source) {
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
if (source) {
// Return an error if trying to set a valid source with an invalid ssrc.
RTC_LOG(LS_ERROR) << "SetLocalSource failed with ssrc " << ssrc;
return false;
}
// The channel likely has gone away, do nothing.
return true;
}
if (source) {
it->second->SetSource(source);
} else {
it->second->ClearSource();
}
return true;
}
bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
std::vector<uint32_t> ssrcs(1, ssrc);
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
default_recv_volume_ = volume;
ssrcs = unsignaled_recv_ssrcs_;
}
for (uint32_t ssrc : ssrcs) {
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
return false;
}
it->second->SetOutputVolume(volume);
RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume
<< " for recv stream with ssrc " << ssrc;
}
return true;
}
bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
return dtmf_payload_type_.has_value() && send_;
}
bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
int event,
int duration) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
if (!CanInsertDtmf()) {
return false;
}
// Figure out which WebRtcAudioSendStream to send the event on.
auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin();
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
if (event < kMinTelephoneEventCode || event > kMaxTelephoneEventCode) {
RTC_LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
return false;
}
RTC_DCHECK_NE(-1, dtmf_payload_freq_);
return it->second->SendTelephoneEvent(*dtmf_payload_type_, dtmf_payload_freq_,
event, duration);
}
void WebRtcVoiceMediaChannel::OnPacketReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
packet_time.timestamp);
if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
return;
}
// Create an unsignaled receive stream for this previously not received ssrc.
// If there already is N unsignaled receive streams, delete the oldest.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
unsignaled_recv_ssrcs_.end(),
ssrc) == unsignaled_recv_ssrcs_.end());
// Add new stream.
StreamParams sp = unsignaled_stream_params_;
sp.ssrcs.push_back(ssrc);
RTC_LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
if (!AddRecvStream(sp)) {
RTC_LOG(LS_WARNING) << "Could not create unsignaled receive stream.";
return;
}
unsignaled_recv_ssrcs_.push_back(ssrc);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.NumOfUnsignaledStreams",
unsignaled_recv_ssrcs_.size(), 1, 100, 101);
// Remove oldest unsignaled stream, if we have too many.
if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
RTC_LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
<< remove_ssrc;
RemoveRecvStream(remove_ssrc);
}
RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());
SetOutputVolume(ssrc, default_recv_volume_);
// The default sink can only be attached to one stream at a time, so we hook
// it up to the *latest* unsignaled stream we've seen, in order to support the
// case where the SSRC of one unsignaled stream changes.
if (default_sink_) {
for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
auto it = recv_streams_.find(drop_ssrc);
it->second->SetRawAudioSink(nullptr);
}
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
new ProxySink(default_sink_.get()));
SetRawAudioSink(ssrc, std::move(proxy_sink));
}
delivery_result = call_->Receiver()->DeliverPacket(
webrtc::MediaType::AUDIO, *packet, packet_time.timestamp);
RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Forward packet to Call as well.
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
packet_time.timestamp);
}
void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
call_->GetTransportControllerSend()->OnNetworkRouteChanged(transport_name,
network_route);
call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
network_route.packet_overhead);
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
it->second->SetMuted(muted);
// TODO(solenberg):
// We set the AGC to mute state only when all the channels are muted.
// This implementation is not ideal, instead we should signal the AGC when
// the mic channel is muted/unmuted. We can't do it today because there
// is no good way to know which stream is mapping to the mic channel.
bool all_muted = muted;
for (const auto& kv : send_streams_) {
all_muted = all_muted && kv.second->muted();
}
engine()->apm()->set_output_will_be_muted(all_muted);
return true;
}
bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate.";
max_send_bitrate_bps_ = bps;
bool success = true;
for (const auto& kv : send_streams_) {
if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) {
success = false;
}
}
return success;
}
void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
call_->SignalChannelNetworkState(
webrtc::MediaType::AUDIO,
ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(info);
// Get SSRC and stats for each sender.
RTC_DCHECK_EQ(info->senders.size(), 0U);
for (const auto& stream : send_streams_) {
webrtc::AudioSendStream::Stats stats =
stream.second->GetStats(recv_streams_.size() > 0);
VoiceSenderInfo sinfo;
sinfo.add_ssrc(stats.local_ssrc);
sinfo.bytes_sent = stats.bytes_sent;
sinfo.packets_sent = stats.packets_sent;
sinfo.packets_lost = stats.packets_lost;
sinfo.fraction_lost = stats.fraction_lost;
sinfo.codec_name = stats.codec_name;
sinfo.codec_payload_type = stats.codec_payload_type;
sinfo.ext_seqnum = stats.ext_seqnum;
sinfo.jitter_ms = stats.jitter_ms;
sinfo.rtt_ms = stats.rtt_ms;
sinfo.audio_level = stats.audio_level;
sinfo.total_input_energy = stats.total_input_energy;
sinfo.total_input_duration = stats.total_input_duration;
sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
sinfo.ana_statistics = stats.ana_statistics;
sinfo.apm_statistics = stats.apm_statistics;
info->senders.push_back(sinfo);
}
// Get SSRC and stats for each receiver.
RTC_DCHECK_EQ(info->receivers.size(), 0U);
for (const auto& stream : recv_streams_) {
uint32_t ssrc = stream.first;
// When SSRCs are unsignaled, there's only one audio MediaStreamTrack, but
// multiple RTP streams can be received over time (if the SSRC changes for
// whatever reason). We only want the RTCMediaStreamTrackStats to represent
// the stats for the most recent stream (the one whose audio is actually
// routed to the MediaStreamTrack), so here we ignore any unsignaled SSRCs
// except for the most recent one (last in the vector). This is somewhat of
// a hack, and means you don't get *any* stats for these inactive streams,
// but it's slightly better than the previous behavior, which was "highest
// SSRC wins".
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
if (!unsignaled_recv_ssrcs_.empty()) {
auto end_it = --unsignaled_recv_ssrcs_.end();
if (std::find(unsignaled_recv_ssrcs_.begin(), end_it, ssrc) != end_it) {
continue;
}
}
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
VoiceReceiverInfo rinfo;
rinfo.add_ssrc(stats.remote_ssrc);
rinfo.bytes_rcvd = stats.bytes_rcvd;
rinfo.packets_rcvd = stats.packets_rcvd;
rinfo.packets_lost = stats.packets_lost;
rinfo.fraction_lost = stats.fraction_lost;
rinfo.codec_name = stats.codec_name;
rinfo.codec_payload_type = stats.codec_payload_type;
rinfo.ext_seqnum = stats.ext_seqnum;
rinfo.jitter_ms = stats.jitter_ms;
rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
rinfo.delay_estimate_ms = stats.delay_estimate_ms;
rinfo.audio_level = stats.audio_level;
rinfo.total_output_energy = stats.total_output_energy;
rinfo.total_samples_received = stats.total_samples_received;
rinfo.total_output_duration = stats.total_output_duration;
rinfo.concealed_samples = stats.concealed_samples;
rinfo.concealment_events = stats.concealment_events;
rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds;
rinfo.expand_rate = stats.expand_rate;
rinfo.speech_expand_rate = stats.speech_expand_rate;
rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
rinfo.secondary_discarded_rate = stats.secondary_discarded_rate;
rinfo.accelerate_rate = stats.accelerate_rate;
rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
rinfo.decoding_calls_to_silence_generator =
stats.decoding_calls_to_silence_generator;
rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
rinfo.decoding_normal = stats.decoding_normal;
rinfo.decoding_plc = stats.decoding_plc;
rinfo.decoding_cng = stats.decoding_cng;
rinfo.decoding_plc_cng = stats.decoding_plc_cng;
rinfo.decoding_muted_output = stats.decoding_muted_output;
rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
info->receivers.push_back(rinfo);
}
// Get codec info
for (const AudioCodec& codec : send_codecs_) {
webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
info->send_codecs.insert(
std::make_pair(codec_params.payload_type, std::move(codec_params)));
}
for (const AudioCodec& codec : recv_codecs_) {
webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
info->receive_codecs.insert(
std::make_pair(codec_params.payload_type, std::move(codec_params)));
}
return true;
}
void WebRtcVoiceMediaChannel::SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:"
<< ssrc << " " << (sink ? "(ptr)" : "NULL");
if (ssrc == 0) {
if (!unsignaled_recv_ssrcs_.empty()) {
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
sink ? new ProxySink(sink.get()) : nullptr);
SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
}
default_sink_ = std::move(sink);
return;
}
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc;
return;
}
it->second->SetRawAudioSink(std::move(sink));
}
std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources(
uint32_t ssrc) const {
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_ERROR) << "Attempting to get contributing sources for SSRC:"
<< ssrc << " which doesn't exist.";
return std::vector<webrtc::RtpSource>();
}
return it->second->GetSources();
}
bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream(
uint32_t ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = std::find(unsignaled_recv_ssrcs_.begin(),
unsignaled_recv_ssrcs_.end(), ssrc);
if (it != unsignaled_recv_ssrcs_.end()) {
unsignaled_recv_ssrcs_.erase(it);
return true;
}
return false;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE
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