diff options
Diffstat (limited to 'talk/media/webrtc/webrtcvoiceengine.cc')
| -rw-r--r-- | talk/media/webrtc/webrtcvoiceengine.cc | 1823 |
1 files changed, 715 insertions, 1108 deletions
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc index 27ca1deb2d..9192b72539 100644 --- a/talk/media/webrtc/webrtcvoiceengine.cc +++ b/talk/media/webrtc/webrtcvoiceengine.cc @@ -42,7 +42,10 @@ #include "talk/media/base/audiorenderer.h" #include "talk/media/base/constants.h" #include "talk/media/base/streamparams.h" +#include "talk/media/webrtc/webrtcmediaengine.h" #include "talk/media/webrtc/webrtcvoe.h" +#include "webrtc/audio/audio_sink.h" +#include "webrtc/base/arraysize.h" #include "webrtc/base/base64.h" #include "webrtc/base/byteorder.h" #include "webrtc/base/common.h" @@ -52,53 +55,26 @@ #include "webrtc/base/stringutils.h" #include "webrtc/call/rtc_event_log.h" #include "webrtc/common.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/system_wrappers/include/field_trial.h" +#include "webrtc/system_wrappers/include/trace.h" namespace cricket { namespace { -const int kMaxNumPacketSize = 6; -struct CodecPref { - const char* name; - int clockrate; - int channels; - int payload_type; - bool is_multi_rate; - int packet_sizes_ms[kMaxNumPacketSize]; -}; -// Note: keep the supported packet sizes in ascending order. -const CodecPref kCodecPrefs[] = { - { kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } }, - { kIsacCodecName, 16000, 1, 103, true, { 30, 60 } }, - { kIsacCodecName, 32000, 1, 104, true, { 30 } }, - // G722 should be advertised as 8000 Hz because of the RFC "bug". - { kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } }, - { kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } }, - { kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } }, - { kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } }, - { kCnCodecName, 32000, 1, 106, false, { } }, - { kCnCodecName, 16000, 1, 105, false, { } }, - { kCnCodecName, 8000, 1, 13, false, { } }, - { kRedCodecName, 8000, 1, 127, false, { } }, - { kDtmfCodecName, 8000, 1, 126, false, { } }, -}; +const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo | + webrtc::kTraceWarning | webrtc::kTraceError | + webrtc::kTraceCritical; +const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo | + webrtc::kTraceInfo; -// For Linux/Mac, using the default device is done by specifying index 0 for -// VoE 4.0 and not -1 (which was the case for VoE 3.5). -// // On Windows Vista and newer, Microsoft introduced the concept of "Default // Communications Device". This means that there are two types of default // devices (old Wave Audio style default and Default Communications Device). // // On Windows systems which only support Wave Audio style default, uses either // -1 or 0 to select the default device. -// -// On Windows systems which support both "Default Communication Device" and -// old Wave Audio style default, use -1 for Default Communications Device and -// -2 for Wave Audio style default, which is what we want to use for clips. -// It's not clear yet whether the -2 index is handled properly on other OSes. - #ifdef WIN32 const int kDefaultAudioDeviceId = -1; #else @@ -150,6 +126,12 @@ const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump"; const char kAecDumpByAudioOptionFilename[] = "audio.aecdump"; #endif +// Constants from voice_engine_defines.h. +const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1) +const int kMaxTelephoneEventCode = 255; +const int kMinTelephoneEventDuration = 100; +const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16 + bool ValidateStreamParams(const StreamParams& sp) { if (sp.ssrcs.empty()) { LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); @@ -177,32 +159,6 @@ std::string ToString(const webrtc::CodecInst& codec) { return ss.str(); } -void LogMultiline(rtc::LoggingSeverity sev, char* text) { - const char* delim = "\r\n"; - for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) { - LOG_V(sev) << tok; - } -} - -// Severity is an integer because it comes is assumed to be from command line. -int SeverityToFilter(int severity) { - int filter = webrtc::kTraceNone; - switch (severity) { - case rtc::LS_VERBOSE: - filter |= webrtc::kTraceAll; - FALLTHROUGH(); - case rtc::LS_INFO: - filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo); - FALLTHROUGH(); - case rtc::LS_WARNING: - filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning); - FALLTHROUGH(); - case rtc::LS_ERROR: - filter |= (webrtc::kTraceError | webrtc::kTraceCritical); - } - return filter; -} - bool IsCodec(const AudioCodec& codec, const char* ref_name) { return (_stricmp(codec.name.c_str(), ref_name) == 0); } @@ -211,19 +167,9 @@ bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) { return (_stricmp(codec.plname, ref_name) == 0); } -bool IsCodecMultiRate(const webrtc::CodecInst& codec) { - for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) { - if (IsCodec(codec, kCodecPrefs[i].name) && - kCodecPrefs[i].clockrate == codec.plfreq) { - return kCodecPrefs[i].is_multi_rate; - } - } - return false; -} - bool FindCodec(const std::vector<AudioCodec>& codecs, - const AudioCodec& codec, - AudioCodec* found_codec) { + const AudioCodec& codec, + AudioCodec* found_codec) { for (const AudioCodec& c : codecs) { if (c.Matches(codec)) { if (found_codec != NULL) { @@ -253,38 +199,8 @@ bool IsNackEnabled(const AudioCodec& codec) { kParamValueEmpty)); } -int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) { - int selected_packet_size_ms = codec_pref.packet_sizes_ms[0]; - for (int packet_size_ms : codec_pref.packet_sizes_ms) { - if (packet_size_ms && packet_size_ms <= ptime_ms) { - selected_packet_size_ms = packet_size_ms; - } - } - return selected_packet_size_ms; -} - -// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec -// pacsize if it's valid, or we will pick the next smallest value we support. -// TODO(Brave): Query supported packet sizes from ACM when the API is ready. -bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) { - for (const CodecPref& codec_pref : kCodecPrefs) { - if ((IsCodec(*codec, codec_pref.name) && - codec_pref.clockrate == codec->plfreq) || - IsCodec(*codec, kG722CodecName)) { - int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms); - if (packet_size_ms) { - // Convert unit from milli-seconds to samples. - codec->pacsize = (codec->plfreq / 1000) * packet_size_ms; - return true; - } - } - } - return false; -} - // Return true if codec.params[feature] == "1", false otherwise. -bool IsCodecFeatureEnabled(const AudioCodec& codec, - const char* feature) { +bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) { int value; return codec.GetParam(feature, &value) && value == 1; } @@ -351,109 +267,29 @@ void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec, voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate); } -// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC -// which says that G722 should be advertised as 8 kHz although it is a 16 kHz -// codec. -void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) { - if (IsCodec(*voe_codec, kG722CodecName)) { - // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine - // has changed, and this special case is no longer needed. - RTC_DCHECK(voe_codec->plfreq != new_plfreq); - voe_codec->plfreq = new_plfreq; - } -} - -// Gets the default set of options applied to the engine. Historically, these -// were supplied as a combination of flags from the channel manager (ec, agc, -// ns, and highpass) and the rest hardcoded in InitInternal. -AudioOptions GetDefaultEngineOptions() { - AudioOptions options; - options.echo_cancellation.Set(true); - options.auto_gain_control.Set(true); - options.noise_suppression.Set(true); - options.highpass_filter.Set(true); - options.stereo_swapping.Set(false); - options.audio_jitter_buffer_max_packets.Set(50); - options.audio_jitter_buffer_fast_accelerate.Set(false); - options.typing_detection.Set(true); - options.adjust_agc_delta.Set(0); - options.experimental_agc.Set(false); - options.extended_filter_aec.Set(false); - options.delay_agnostic_aec.Set(false); - options.experimental_ns.Set(false); - options.aec_dump.Set(false); - return options; -} - -std::string GetEnableString(bool enable) { - return enable ? "enable" : "disable"; -} -} // namespace { - -WebRtcVoiceEngine::WebRtcVoiceEngine() - : voe_wrapper_(new VoEWrapper()), - tracing_(new VoETraceWrapper()), - adm_(NULL), - log_filter_(SeverityToFilter(kDefaultLogSeverity)), - is_dumping_aec_(false) { - Construct(); -} - -WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper, - VoETraceWrapper* tracing) - : voe_wrapper_(voe_wrapper), - tracing_(tracing), - adm_(NULL), - log_filter_(SeverityToFilter(kDefaultLogSeverity)), - is_dumping_aec_(false) { - Construct(); -} - -void WebRtcVoiceEngine::Construct() { - SetTraceFilter(log_filter_); - initialized_ = false; - LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; - SetTraceOptions(""); - if (tracing_->SetTraceCallback(this) == -1) { - LOG_RTCERR0(SetTraceCallback); - } - if (voe_wrapper_->base()->RegisterVoiceEngineObserver(*this) == -1) { - LOG_RTCERR0(RegisterVoiceEngineObserver); - } - // Clear the default agc state. - memset(&default_agc_config_, 0, sizeof(default_agc_config_)); - - // Load our audio codec list. - ConstructCodecs(); - - // Load our RTP Header extensions. - rtp_header_extensions_.push_back( - RtpHeaderExtension(kRtpAudioLevelHeaderExtension, - kRtpAudioLevelHeaderExtensionDefaultId)); - rtp_header_extensions_.push_back( - RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, - kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); - if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") { - rtp_header_extensions_.push_back(RtpHeaderExtension( - kRtpTransportSequenceNumberHeaderExtension, - kRtpTransportSequenceNumberHeaderExtensionDefaultId)); - } - options_ = GetDefaultEngineOptions(); +webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) { + webrtc::AudioState::Config config; + config.voice_engine = voe_wrapper->engine(); + return config; } -void WebRtcVoiceEngine::ConstructCodecs() { - LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; - int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); - for (int i = 0; i < ncodecs; ++i) { - webrtc::CodecInst voe_codec; - if (GetVoeCodec(i, &voe_codec)) { +class WebRtcVoiceCodecs final { + public: + // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec + // list and add a test which verifies VoE supports the listed codecs. + static std::vector<AudioCodec> SupportedCodecs() { + LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; + std::vector<AudioCodec> result; + for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) { + // Change the sample rate of G722 to 8000 to match SDP. + MaybeFixupG722(&voe_codec, 8000); // Skip uncompressed formats. if (IsCodec(voe_codec, kL16CodecName)) { continue; } const CodecPref* pref = NULL; - for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) { + for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) { if (IsCodec(voe_codec, kCodecPrefs[j].name) && kCodecPrefs[j].clockrate == voe_codec.plfreq && kCodecPrefs[j].channels == voe_codec.channels) { @@ -465,9 +301,10 @@ void WebRtcVoiceEngine::ConstructCodecs() { if (pref) { // Use the payload type that we've configured in our pref table; // use the offset in our pref table to determine the sort order. - AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq, - voe_codec.rate, voe_codec.channels, - ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs)); + AudioCodec codec( + pref->payload_type, voe_codec.plname, voe_codec.plfreq, + voe_codec.rate, voe_codec.channels, + static_cast<int>(arraysize(kCodecPrefs)) - (pref - kCodecPrefs)); LOG(LS_INFO) << ToString(codec); if (IsCodec(codec, kIsacCodecName)) { // Indicate auto-bitrate in signaling. @@ -488,40 +325,183 @@ void WebRtcVoiceEngine::ConstructCodecs() { // TODO(hellner): Add ptime, sprop-stereo, and stereo // when they can be set to values other than the default. } - codecs_.push_back(codec); + result.push_back(codec); } else { LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec); } } + // Make sure they are in local preference order. + std::sort(result.begin(), result.end(), &AudioCodec::Preferable); + return result; + } + + static bool ToCodecInst(const AudioCodec& in, + webrtc::CodecInst* out) { + for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) { + // Change the sample rate of G722 to 8000 to match SDP. + MaybeFixupG722(&voe_codec, 8000); + AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq, + voe_codec.rate, voe_codec.channels, 0); + bool multi_rate = IsCodecMultiRate(voe_codec); + // Allow arbitrary rates for ISAC to be specified. + if (multi_rate) { + // Set codec.bitrate to 0 so the check for codec.Matches() passes. + codec.bitrate = 0; + } + if (codec.Matches(in)) { + if (out) { + // Fixup the payload type. + voe_codec.pltype = in.id; + + // Set bitrate if specified. + if (multi_rate && in.bitrate != 0) { + voe_codec.rate = in.bitrate; + } + + // Reset G722 sample rate to 16000 to match WebRTC. + MaybeFixupG722(&voe_codec, 16000); + + // Apply codec-specific settings. + if (IsCodec(codec, kIsacCodecName)) { + // If ISAC and an explicit bitrate is not specified, + // enable auto bitrate adjustment. + voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1; + } + *out = voe_codec; + } + return true; + } + } + return false; } - // Make sure they are in local preference order. - std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable); -} -bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) { - if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) { + static bool IsCodecMultiRate(const webrtc::CodecInst& codec) { + for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) { + if (IsCodec(codec, kCodecPrefs[i].name) && + kCodecPrefs[i].clockrate == codec.plfreq) { + return kCodecPrefs[i].is_multi_rate; + } + } return false; } - // Change the sample rate of G722 to 8000 to match SDP. - MaybeFixupG722(codec, 8000); - return true; + + // If the AudioCodec param kCodecParamPTime is set, then we will set it to + // codec pacsize if it's valid, or we will pick the next smallest value we + // support. + // TODO(Brave): Query supported packet sizes from ACM when the API is ready. + static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) { + for (const CodecPref& codec_pref : kCodecPrefs) { + if ((IsCodec(*codec, codec_pref.name) && + codec_pref.clockrate == codec->plfreq) || + IsCodec(*codec, kG722CodecName)) { + int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms); + if (packet_size_ms) { + // Convert unit from milli-seconds to samples. + codec->pacsize = (codec->plfreq / 1000) * packet_size_ms; + return true; + } + } + } + return false; + } + + private: + static const int kMaxNumPacketSize = 6; + struct CodecPref { + const char* name; + int clockrate; + size_t channels; + int payload_type; + bool is_multi_rate; + int packet_sizes_ms[kMaxNumPacketSize]; + }; + // Note: keep the supported packet sizes in ascending order. + static const CodecPref kCodecPrefs[12]; + + static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) { + int selected_packet_size_ms = codec_pref.packet_sizes_ms[0]; + for (int packet_size_ms : codec_pref.packet_sizes_ms) { + if (packet_size_ms && packet_size_ms <= ptime_ms) { + selected_packet_size_ms = packet_size_ms; + } + } + return selected_packet_size_ms; + } + + // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC + // which says that G722 should be advertised as 8 kHz although it is a 16 kHz + // codec. + static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) { + if (IsCodec(*voe_codec, kG722CodecName)) { + // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine + // has changed, and this special case is no longer needed. + RTC_DCHECK(voe_codec->plfreq != new_plfreq); + voe_codec->plfreq = new_plfreq; + } + } +}; + +const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = { + { kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } }, + { kIsacCodecName, 16000, 1, 103, true, { 30, 60 } }, + { kIsacCodecName, 32000, 1, 104, true, { 30 } }, + // G722 should be advertised as 8000 Hz because of the RFC "bug". + { kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } }, + { kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } }, + { kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } }, + { kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } }, + { kCnCodecName, 32000, 1, 106, false, { } }, + { kCnCodecName, 16000, 1, 105, false, { } }, + { kCnCodecName, 8000, 1, 13, false, { } }, + { kRedCodecName, 8000, 1, 127, false, { } }, + { kDtmfCodecName, 8000, 1, 126, false, { } }, +}; +} // namespace { + +bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in, + webrtc::CodecInst* out) { + return WebRtcVoiceCodecs::ToCodecInst(in, out); +} + +WebRtcVoiceEngine::WebRtcVoiceEngine() + : voe_wrapper_(new VoEWrapper()), + audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))) { + Construct(); +} + +WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper) + : voe_wrapper_(voe_wrapper) { + Construct(); +} + +void WebRtcVoiceEngine::Construct() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine"; + + signal_thread_checker_.DetachFromThread(); + std::memset(&default_agc_config_, 0, sizeof(default_agc_config_)); + voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true)); + + webrtc::Trace::set_level_filter(kDefaultTraceFilter); + webrtc::Trace::SetTraceCallback(this); + + // Load our audio codec list. + codecs_ = WebRtcVoiceCodecs::SupportedCodecs(); } WebRtcVoiceEngine::~WebRtcVoiceEngine() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; - if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) { - LOG_RTCERR0(DeRegisterVoiceEngineObserver); - } if (adm_) { voe_wrapper_.reset(); adm_->Release(); adm_ = NULL; } - - tracing_->SetTraceCallback(NULL); + webrtc::Trace::SetTraceCallback(nullptr); } bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(worker_thread == rtc::Thread::Current()); LOG(LS_INFO) << "WebRtcVoiceEngine::Init"; bool res = InitInternal(); @@ -535,59 +515,37 @@ bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) { } bool WebRtcVoiceEngine::InitInternal() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // Temporarily turn logging level up for the Init call - int old_filter = log_filter_; - int extended_filter = log_filter_ | SeverityToFilter(rtc::LS_INFO); - SetTraceFilter(extended_filter); - SetTraceOptions(""); - - // Init WebRtc VoiceEngine. + webrtc::Trace::set_level_filter(kElevatedTraceFilter); + LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString(); if (voe_wrapper_->base()->Init(adm_) == -1) { LOG_RTCERR0_EX(Init, voe_wrapper_->error()); - SetTraceFilter(old_filter); return false; } - - SetTraceFilter(old_filter); - SetTraceOptions(log_options_); - - // Log the VoiceEngine version info - char buffer[1024] = ""; - voe_wrapper_->base()->GetVersion(buffer); - LOG(LS_INFO) << "WebRtc VoiceEngine Version:"; - LogMultiline(rtc::LS_INFO, buffer); + webrtc::Trace::set_level_filter(kDefaultTraceFilter); // Save the default AGC configuration settings. This must happen before - // calling SetOptions or the default will be overwritten. + // calling ApplyOptions or the default will be overwritten. if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) { LOG_RTCERR0(GetAgcConfig); return false; } - // Set defaults for options, so that ApplyOptions applies them explicitly - // when we clear option (channel) overrides. External clients can still - // modify the defaults via SetOptions (on the media engine). - if (!SetOptions(GetDefaultEngineOptions())) { - return false; - } - // Print our codec list again for the call diagnostic log LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; for (const AudioCodec& codec : codecs_) { LOG(LS_INFO) << ToString(codec); } - // Disable the DTMF playout when a tone is sent. - // PlayDtmfTone will be used if local playout is needed. - if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) { - LOG_RTCERR1(SetDtmfFeedbackStatus, false); - } + SetDefaultDevices(); initialized_ = true; return true; } void WebRtcVoiceEngine::Terminate() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate"; initialized_ = false; @@ -596,62 +554,81 @@ void WebRtcVoiceEngine::Terminate() { voe_wrapper_->base()->Terminate(); } +rtc::scoped_refptr<webrtc::AudioState> + WebRtcVoiceEngine::GetAudioState() const { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + return audio_state_; +} + VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call, const AudioOptions& options) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); return new WebRtcVoiceMediaChannel(this, options, call); } -bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) { - if (!ApplyOptions(options)) { - return false; - } - options_ = options; - return true; -} - -// AudioOptions defaults are set in InitInternal (for options with corresponding -// MediaEngineInterface flags) and in SetOptions(int) for flagless options. bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString(); - AudioOptions options = options_in; // The options are modified below. + + // Default engine options. + AudioOptions options; + options.echo_cancellation = rtc::Optional<bool>(true); + options.auto_gain_control = rtc::Optional<bool>(true); + options.noise_suppression = rtc::Optional<bool>(true); + options.highpass_filter = rtc::Optional<bool>(true); + options.stereo_swapping = rtc::Optional<bool>(false); + options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50); + options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false); + options.typing_detection = rtc::Optional<bool>(true); + options.adjust_agc_delta = rtc::Optional<int>(0); + options.experimental_agc = rtc::Optional<bool>(false); + options.extended_filter_aec = rtc::Optional<bool>(false); + options.delay_agnostic_aec = rtc::Optional<bool>(false); + options.experimental_ns = rtc::Optional<bool>(false); + options.aec_dump = rtc::Optional<bool>(false); + + // Apply any given options on top. + options.SetAll(options_in); + // kEcConference is AEC with high suppression. webrtc::EcModes ec_mode = webrtc::kEcConference; webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone; webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog; webrtc::NsModes ns_mode = webrtc::kNsHighSuppression; - bool aecm_comfort_noise = false; - if (options.aecm_generate_comfort_noise.Get(&aecm_comfort_noise)) { + if (options.aecm_generate_comfort_noise) { LOG(LS_VERBOSE) << "Comfort noise explicitly set to " - << aecm_comfort_noise << " (default is false)."; + << *options.aecm_generate_comfort_noise + << " (default is false)."; } -#if defined(IOS) +#if defined(WEBRTC_IOS) // On iOS, VPIO provides built-in EC and AGC. - options.echo_cancellation.Set(false); - options.auto_gain_control.Set(false); + options.echo_cancellation = rtc::Optional<bool>(false); + options.auto_gain_control = rtc::Optional<bool>(false); LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead."; #elif defined(ANDROID) ec_mode = webrtc::kEcAecm; #endif -#if defined(IOS) || defined(ANDROID) +#if defined(WEBRTC_IOS) || defined(ANDROID) // Set the AGC mode for iOS as well despite disabling it above, to avoid // unsupported configuration errors from webrtc. agc_mode = webrtc::kAgcFixedDigital; - options.typing_detection.Set(false); - options.experimental_agc.Set(false); - options.extended_filter_aec.Set(false); - options.experimental_ns.Set(false); + options.typing_detection = rtc::Optional<bool>(false); + options.experimental_agc = rtc::Optional<bool>(false); + options.extended_filter_aec = rtc::Optional<bool>(false); + options.experimental_ns = rtc::Optional<bool>(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(IOS) - if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) { +#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.Set(true); - options.extended_filter_aec.Set(true); + options.echo_cancellation = rtc::Optional<bool>(true); + options.extended_filter_aec = rtc::Optional<bool>(true); ec_mode = webrtc::kEcConference; } } @@ -659,8 +636,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing(); - bool echo_cancellation = false; - if (options.echo_cancellation.Get(&echo_cancellation)) { + 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 @@ -671,63 +647,61 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { // overriding it. Enable/Disable it according to the echo_cancellation // audio option. const bool enable_built_in_aec = - echo_cancellation && !use_delay_agnostic_aec; + *options.echo_cancellation && !use_delay_agnostic_aec; if (voe_wrapper_->hw()->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.Set(false); - echo_cancellation = false; + options.echo_cancellation = rtc::Optional<bool>(false); LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead"; } } - if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) { - LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode); + if (voep->SetEcStatus(*options.echo_cancellation, ec_mode) == -1) { + LOG_RTCERR2(SetEcStatus, *options.echo_cancellation, ec_mode); return false; } else { - LOG(LS_INFO) << "Echo control set to " << echo_cancellation + LOG(LS_INFO) << "Echo control set to " << *options.echo_cancellation << " with mode " << ec_mode; } #if !defined(ANDROID) // TODO(ajm): Remove the error return on Android from webrtc. - if (voep->SetEcMetricsStatus(echo_cancellation) == -1) { - LOG_RTCERR1(SetEcMetricsStatus, echo_cancellation); + if (voep->SetEcMetricsStatus(*options.echo_cancellation) == -1) { + LOG_RTCERR1(SetEcMetricsStatus, *options.echo_cancellation); return false; } #endif if (ec_mode == webrtc::kEcAecm) { - if (voep->SetAecmMode(aecm_mode, aecm_comfort_noise) != 0) { - LOG_RTCERR2(SetAecmMode, aecm_mode, aecm_comfort_noise); + bool cn = options.aecm_generate_comfort_noise.value_or(false); + if (voep->SetAecmMode(aecm_mode, cn) != 0) { + LOG_RTCERR2(SetAecmMode, aecm_mode, cn); return false; } } } - bool auto_gain_control = false; - if (options.auto_gain_control.Get(&auto_gain_control)) { + if (options.auto_gain_control) { const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable(); if (built_in_agc) { - if (voe_wrapper_->hw()->EnableBuiltInAGC(auto_gain_control) == 0 && - auto_gain_control) { + if (voe_wrapper_->hw()->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.Set(false); - auto_gain_control = false; + options.auto_gain_control = rtc::Optional<bool>(false); LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead"; } } - if (voep->SetAgcStatus(auto_gain_control, agc_mode) == -1) { - LOG_RTCERR2(SetAgcStatus, auto_gain_control, agc_mode); + if (voep->SetAgcStatus(*options.auto_gain_control, agc_mode) == -1) { + LOG_RTCERR2(SetAgcStatus, *options.auto_gain_control, agc_mode); return false; } else { - LOG(LS_INFO) << "Auto gain set to " << auto_gain_control << " with mode " - << agc_mode; + LOG(LS_INFO) << "Auto gain set to " << *options.auto_gain_control + << " with mode " << agc_mode; } } - if (options.tx_agc_target_dbov.IsSet() || - options.tx_agc_digital_compression_gain.IsSet() || - options.tx_agc_limiter.IsSet()) { + 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. @@ -736,15 +710,13 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { // Also, if we don't update default_agc_config_, then adjust_agc_delta // would be an offset from the original values, and not whatever was set // explicitly. - default_agc_config_.targetLeveldBOv = - options.tx_agc_target_dbov.GetWithDefaultIfUnset( - default_agc_config_.targetLeveldBOv); + 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.GetWithDefaultIfUnset( + options.tx_agc_digital_compression_gain.value_or( default_agc_config_.digitalCompressionGaindB); default_agc_config_.limiterEnable = - options.tx_agc_limiter.GetWithDefaultIfUnset( - default_agc_config_.limiterEnable); + options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable); if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) { LOG_RTCERR3(SetAgcConfig, default_agc_config_.targetLeveldBOv, @@ -754,84 +726,79 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { } } - bool noise_suppression = false; - if (options.noise_suppression.Get(&noise_suppression)) { + if (options.noise_suppression) { const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable(); if (built_in_ns) { - if (voe_wrapper_->hw()->EnableBuiltInNS(noise_suppression) == 0 && - noise_suppression) { + if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) == + 0 && + *options.noise_suppression) { // Disable internal software NS if built-in NS is enabled, // i.e., replace the software NS with the built-in NS. - options.noise_suppression.Set(false); - noise_suppression = false; + options.noise_suppression = rtc::Optional<bool>(false); LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead"; } } - if (voep->SetNsStatus(noise_suppression, ns_mode) == -1) { - LOG_RTCERR2(SetNsStatus, noise_suppression, ns_mode); + if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) { + LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode); return false; } else { - LOG(LS_INFO) << "Noise suppression set to " << noise_suppression + LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression << " with mode " << ns_mode; } } - bool highpass_filter; - if (options.highpass_filter.Get(&highpass_filter)) { - LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter; - if (voep->EnableHighPassFilter(highpass_filter) == -1) { - LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter); + if (options.highpass_filter) { + LOG(LS_INFO) << "High pass filter enabled? " << *options.highpass_filter; + if (voep->EnableHighPassFilter(*options.highpass_filter) == -1) { + LOG_RTCERR1(SetHighpassFilterStatus, *options.highpass_filter); return false; } } - bool stereo_swapping; - if (options.stereo_swapping.Get(&stereo_swapping)) { - LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping; - voep->EnableStereoChannelSwapping(stereo_swapping); - if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) { - LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping); + if (options.stereo_swapping) { + LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping; + voep->EnableStereoChannelSwapping(*options.stereo_swapping); + if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) { + LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping); return false; } } - int audio_jitter_buffer_max_packets; - if (options.audio_jitter_buffer_max_packets.Get( - &audio_jitter_buffer_max_packets)) { - LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets; + if (options.audio_jitter_buffer_max_packets) { + LOG(LS_INFO) << "NetEq capacity is " + << *options.audio_jitter_buffer_max_packets; voe_config_.Set<webrtc::NetEqCapacityConfig>( - new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets)); + new webrtc::NetEqCapacityConfig( + *options.audio_jitter_buffer_max_packets)); } - bool audio_jitter_buffer_fast_accelerate; - if (options.audio_jitter_buffer_fast_accelerate.Get( - &audio_jitter_buffer_fast_accelerate)) { - LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate; + if (options.audio_jitter_buffer_fast_accelerate) { + LOG(LS_INFO) << "NetEq fast mode? " + << *options.audio_jitter_buffer_fast_accelerate; voe_config_.Set<webrtc::NetEqFastAccelerate>( - new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate)); + new webrtc::NetEqFastAccelerate( + *options.audio_jitter_buffer_fast_accelerate)); } - bool typing_detection; - if (options.typing_detection.Get(&typing_detection)) { - LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection; - if (voep->SetTypingDetectionStatus(typing_detection) == -1) { + if (options.typing_detection) { + LOG(LS_INFO) << "Typing detection is enabled? " + << *options.typing_detection; + if (voep->SetTypingDetectionStatus(*options.typing_detection) == -1) { // In case of error, log the info and continue - LOG_RTCERR1(SetTypingDetectionStatus, typing_detection); + LOG_RTCERR1(SetTypingDetectionStatus, *options.typing_detection); } } - int adjust_agc_delta; - if (options.adjust_agc_delta.Get(&adjust_agc_delta)) { - LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta; - if (!AdjustAgcLevel(adjust_agc_delta)) { + if (options.adjust_agc_delta) { + LOG(LS_INFO) << "Adjust agc delta is " << *options.adjust_agc_delta; + if (!AdjustAgcLevel(*options.adjust_agc_delta)) { return false; } } - bool aec_dump; - if (options.aec_dump.Get(&aec_dump)) { - LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump; - if (aec_dump) + if (options.aec_dump) { + LOG(LS_INFO) << "Aec dump is enabled? " << *options.aec_dump; + if (*options.aec_dump) StartAecDump(kAecDumpByAudioOptionFilename); else StopAecDump(); @@ -839,28 +806,30 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { webrtc::Config config; - delay_agnostic_aec_.SetFrom(options.delay_agnostic_aec); - bool delay_agnostic_aec; - if (delay_agnostic_aec_.Get(&delay_agnostic_aec)) { - LOG(LS_INFO) << "Delay agnostic aec is enabled? " << delay_agnostic_aec; + if (options.delay_agnostic_aec) + delay_agnostic_aec_ = options.delay_agnostic_aec; + if (delay_agnostic_aec_) { + LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_; config.Set<webrtc::DelayAgnostic>( - new webrtc::DelayAgnostic(delay_agnostic_aec)); + new webrtc::DelayAgnostic(*delay_agnostic_aec_)); } - extended_filter_aec_.SetFrom(options.extended_filter_aec); - bool extended_filter; - if (extended_filter_aec_.Get(&extended_filter)) { - LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter; + if (options.extended_filter_aec) { + extended_filter_aec_ = options.extended_filter_aec; + } + if (extended_filter_aec_) { + LOG(LS_INFO) << "Extended filter aec is enabled? " << *extended_filter_aec_; config.Set<webrtc::ExtendedFilter>( - new webrtc::ExtendedFilter(extended_filter)); + new webrtc::ExtendedFilter(*extended_filter_aec_)); } - experimental_ns_.SetFrom(options.experimental_ns); - bool experimental_ns; - if (experimental_ns_.Get(&experimental_ns)) { - LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns; + if (options.experimental_ns) { + experimental_ns_ = options.experimental_ns; + } + if (experimental_ns_) { + LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_; config.Set<webrtc::ExperimentalNs>( - new webrtc::ExperimentalNs(experimental_ns)); + new webrtc::ExperimentalNs(*experimental_ns_)); } // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine @@ -870,167 +839,58 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { audioproc->SetExtraOptions(config); } - uint32_t recording_sample_rate; - if (options.recording_sample_rate.Get(&recording_sample_rate)) { - LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate; - if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) { - LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate); + if (options.recording_sample_rate) { + LOG(LS_INFO) << "Recording sample rate is " + << *options.recording_sample_rate; + if (voe_wrapper_->hw()->SetRecordingSampleRate( + *options.recording_sample_rate)) { + LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate); } } - uint32_t playout_sample_rate; - if (options.playout_sample_rate.Get(&playout_sample_rate)) { - LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate; - if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) { - LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate); + if (options.playout_sample_rate) { + LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate; + if (voe_wrapper_->hw()->SetPlayoutSampleRate( + *options.playout_sample_rate)) { + LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate); } } return true; } -// TODO(juberti): Refactor this so that the core logic can be used to set the -// soundclip device. At that time, reinstate the soundclip pause/resume code. -bool WebRtcVoiceEngine::SetDevices(const Device* in_device, - const Device* out_device) { -#if !defined(IOS) - int in_id = in_device ? rtc::FromString<int>(in_device->id) : - kDefaultAudioDeviceId; - int out_id = out_device ? rtc::FromString<int>(out_device->id) : - kDefaultAudioDeviceId; - // The device manager uses -1 as the default device, which was the case for - // VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac. -#ifndef WIN32 - if (-1 == in_id) { - in_id = kDefaultAudioDeviceId; - } - if (-1 == out_id) { - out_id = kDefaultAudioDeviceId; - } -#endif - - std::string in_name = (in_id != kDefaultAudioDeviceId) ? - in_device->name : "Default device"; - std::string out_name = (out_id != kDefaultAudioDeviceId) ? - out_device->name : "Default device"; - LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name - << ") and speaker to (id=" << out_id << ", name=" << out_name - << ")"; +void WebRtcVoiceEngine::SetDefaultDevices() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); +#if !defined(WEBRTC_IOS) + int in_id = kDefaultAudioDeviceId; + int out_id = kDefaultAudioDeviceId; + LOG(LS_INFO) << "Setting microphone to (id=" << in_id + << ") and speaker to (id=" << out_id << ")"; - // Must also pause all audio playback and capture. bool ret = true; - for (WebRtcVoiceMediaChannel* channel : channels_) { - if (!channel->PausePlayout()) { - LOG(LS_WARNING) << "Failed to pause playout"; - ret = false; - } - if (!channel->PauseSend()) { - LOG(LS_WARNING) << "Failed to pause send"; - ret = false; - } - } - - // Find the recording device id in VoiceEngine and set recording device. - if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) { + if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) { + LOG_RTCERR1(SetRecordingDevice, in_id); ret = false; } - if (ret) { - if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) { - LOG_RTCERR2(SetRecordingDevice, in_name, in_id); - ret = false; - } - webrtc::AudioProcessing* ap = voe()->base()->audio_processing(); - if (ap) - ap->Initialize(); + webrtc::AudioProcessing* ap = voe()->base()->audio_processing(); + if (ap) { + ap->Initialize(); } - // Find the playout device id in VoiceEngine and set playout device. - if (!FindWebRtcAudioDeviceId(false, out_name, out_id, &out_id)) { - LOG(LS_WARNING) << "Failed to find VoiceEngine device id for " << out_name; + if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) { + LOG_RTCERR1(SetPlayoutDevice, out_id); ret = false; } - if (ret) { - if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) { - LOG_RTCERR2(SetPlayoutDevice, out_name, out_id); - ret = false; - } - } - - // Resume all audio playback and capture. - for (WebRtcVoiceMediaChannel* channel : channels_) { - if (!channel->ResumePlayout()) { - LOG(LS_WARNING) << "Failed to resume playout"; - ret = false; - } - if (!channel->ResumeSend()) { - LOG(LS_WARNING) << "Failed to resume send"; - ret = false; - } - } if (ret) { - LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name - << ") and speaker to (id="<< out_id << " name=" << out_name - << ")"; + LOG(LS_INFO) << "Set microphone to (id=" << in_id + << ") and speaker to (id=" << out_id << ")"; } - - return ret; -#else - return true; -#endif // !IOS -} - -bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId( - bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) { - // In Linux, VoiceEngine uses the same device dev_id as the device manager. -#if defined(LINUX) || defined(ANDROID) - *rtc_id = dev_id; - return true; -#else - // In Windows and Mac, we need to find the VoiceEngine device id by name - // unless the input dev_id is the default device id. - if (kDefaultAudioDeviceId == dev_id) { - *rtc_id = dev_id; - return true; - } - - // Get the number of VoiceEngine audio devices. - int count = 0; - if (is_input) { - if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) { - LOG_RTCERR0(GetNumOfRecordingDevices); - return false; - } - } else { - if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) { - LOG_RTCERR0(GetNumOfPlayoutDevices); - return false; - } - } - - for (int i = 0; i < count; ++i) { - char name[128]; - char guid[128]; - if (is_input) { - voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid); - LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name; - } else { - voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid); - LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name; - } - - std::string webrtc_name(name); - if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) { - *rtc_id = i; - return true; - } - } - LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name; - return false; -#endif +#endif // !WEBRTC_IOS } bool WebRtcVoiceEngine::GetOutputVolume(int* level) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); unsigned int ulevel; if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) { LOG_RTCERR1(GetSpeakerVolume, level); @@ -1041,6 +901,7 @@ bool WebRtcVoiceEngine::GetOutputVolume(int* level) { } bool WebRtcVoiceEngine::SetOutputVolume(int level) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(level >= 0 && level <= 255); if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) { LOG_RTCERR1(SetSpeakerVolume, level); @@ -1050,136 +911,36 @@ bool WebRtcVoiceEngine::SetOutputVolume(int level) { } int WebRtcVoiceEngine::GetInputLevel() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); unsigned int ulevel; return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ? static_cast<int>(ulevel) : -1; } const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { + RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); return codecs_; } -bool WebRtcVoiceEngine::FindCodec(const AudioCodec& in) { - return FindWebRtcCodec(in, NULL); -} - -// Get the VoiceEngine codec that matches |in|, with the supplied settings. -bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in, - webrtc::CodecInst* out) { - int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); - for (int i = 0; i < ncodecs; ++i) { - webrtc::CodecInst voe_codec; - if (GetVoeCodec(i, &voe_codec)) { - AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq, - voe_codec.rate, voe_codec.channels, 0); - bool multi_rate = IsCodecMultiRate(voe_codec); - // Allow arbitrary rates for ISAC to be specified. - if (multi_rate) { - // Set codec.bitrate to 0 so the check for codec.Matches() passes. - codec.bitrate = 0; - } - if (codec.Matches(in)) { - if (out) { - // Fixup the payload type. - voe_codec.pltype = in.id; - - // Set bitrate if specified. - if (multi_rate && in.bitrate != 0) { - voe_codec.rate = in.bitrate; - } - - // Reset G722 sample rate to 16000 to match WebRTC. - MaybeFixupG722(&voe_codec, 16000); - - // Apply codec-specific settings. - if (IsCodec(codec, kIsacCodecName)) { - // If ISAC and an explicit bitrate is not specified, - // enable auto bitrate adjustment. - voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1; - } - *out = voe_codec; - } - return true; - } - } - } - return false; -} -const std::vector<RtpHeaderExtension>& -WebRtcVoiceEngine::rtp_header_extensions() const { - return rtp_header_extensions_; -} - -void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) { - // if min_sev == -1, we keep the current log level. - if (min_sev >= 0) { - SetTraceFilter(SeverityToFilter(min_sev)); - } - log_options_ = filter; - SetTraceOptions(initialized_ ? log_options_ : ""); +RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { + RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); + RtpCapabilities capabilities; + capabilities.header_extensions.push_back(RtpHeaderExtension( + kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId)); + capabilities.header_extensions.push_back( + RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, + kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); + return capabilities; } int WebRtcVoiceEngine::GetLastEngineError() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); return voe_wrapper_->error(); } -void WebRtcVoiceEngine::SetTraceFilter(int filter) { - log_filter_ = filter; - tracing_->SetTraceFilter(filter); -} - -// We suppport three different logging settings for VoiceEngine: -// 1. Observer callback that goes into talk diagnostic logfile. -// Use --logfile and --loglevel -// -// 2. Encrypted VoiceEngine log for debugging VoiceEngine. -// Use --voice_loglevel --voice_logfilter "tracefile file_name" -// -// 3. EC log and dump for debugging QualityEngine. -// Use --voice_loglevel --voice_logfilter "recordEC file_name" -// -// For more details see: "https://sites.google.com/a/google.com/wavelet/Home/ -// Magic-Flute--RTC-Engine-/Magic-Flute-Command-Line-Parameters" -void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) { - // Set encrypted trace file. - std::vector<std::string> opts; - rtc::tokenize(options, ' ', '"', '"', &opts); - std::vector<std::string>::iterator tracefile = - std::find(opts.begin(), opts.end(), "tracefile"); - if (tracefile != opts.end() && ++tracefile != opts.end()) { - // Write encrypted debug output (at same loglevel) to file - // EncryptedTraceFile no longer supported. - if (tracing_->SetTraceFile(tracefile->c_str()) == -1) { - LOG_RTCERR1(SetTraceFile, *tracefile); - } - } - - // Allow trace options to override the trace filter. We default - // it to log_filter_ (as a translation of libjingle log levels) - // elsewhere, but this allows clients to explicitly set webrtc - // log levels. - std::vector<std::string>::iterator tracefilter = - std::find(opts.begin(), opts.end(), "tracefilter"); - if (tracefilter != opts.end() && ++tracefilter != opts.end()) { - if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) { - LOG_RTCERR1(SetTraceFilter, *tracefilter); - } - } - - // Set AEC dump file - std::vector<std::string>::iterator recordEC = - std::find(opts.begin(), opts.end(), "recordEC"); - if (recordEC != opts.end()) { - ++recordEC; - if (recordEC != opts.end()) - StartAecDump(recordEC->c_str()); - else - StopAecDump(); - } -} - void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, int length) { + // Note: This callback can happen on any thread! rtc::LoggingSeverity sev = rtc::LS_VERBOSE; if (level == webrtc::kTraceError || level == webrtc::kTraceCritical) sev = rtc::LS_ERROR; @@ -1201,34 +962,24 @@ void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, } } -void WebRtcVoiceEngine::CallbackOnError(int channel_id, int err_code) { - RTC_DCHECK(channel_id == -1); - LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel " - << channel_id << "."; - rtc::CritScope lock(&channels_cs_); - for (WebRtcVoiceMediaChannel* channel : channels_) { - channel->OnError(err_code); - } -} - void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) { - RTC_DCHECK(channel != NULL); - rtc::CritScope lock(&channels_cs_); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(channel); channels_.push_back(channel); } void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) { - rtc::CritScope lock(&channels_cs_); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); auto it = std::find(channels_.begin(), channels_.end(), channel); - if (it != channels_.end()) { - channels_.erase(it); - } + RTC_DCHECK(it != channels_.end()); + channels_.erase(it); } // Adjusts the default AGC target level by the specified delta. // NB: If we start messing with other config fields, we'll want // to save the current webrtc::AgcConfig as well. bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); webrtc::AgcConfig config = default_agc_config_; config.targetLeveldBOv -= delta; @@ -1244,6 +995,7 @@ bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) { } bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (initialized_) { LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init."; return false; @@ -1260,6 +1012,7 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) { } bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); if (!aec_dump_file_stream) { LOG(LS_ERROR) << "Could not open AEC dump file stream."; @@ -1279,6 +1032,7 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { } void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (!is_dumping_aec_) { // Start dumping AEC when we are not dumping. if (voe_wrapper_->processing()->StartDebugRecording( @@ -1291,6 +1045,7 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { } void WebRtcVoiceEngine::StopAecDump() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (is_dumping_aec_) { // Stop dumping AEC when we are dumping. if (voe_wrapper_->processing()->StopDebugRecording() != @@ -1302,14 +1057,17 @@ void WebRtcVoiceEngine::StopAecDump() { } bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); return voe_wrapper_->codec()->GetEventLog()->StartLogging(file); } void WebRtcVoiceEngine::StopRtcEventLog() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); voe_wrapper_->codec()->GetEventLog()->StopLogging(); } int WebRtcVoiceEngine::CreateVoEChannel() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); return voe_wrapper_->base()->CreateChannel(voe_config_); } @@ -1317,33 +1075,61 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream : public AudioRenderer::Sink { public: WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport, - uint32_t ssrc, webrtc::Call* call) - : channel_(ch), - voe_audio_transport_(voe_audio_transport), - call_(call) { + uint32_t ssrc, const std::string& c_name, + const std::vector<webrtc::RtpExtension>& extensions, + webrtc::Call* call) + : voe_audio_transport_(voe_audio_transport), + call_(call), + config_(nullptr) { RTC_DCHECK_GE(ch, 0); // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore: // RTC_DCHECK(voe_audio_transport); RTC_DCHECK(call); audio_capture_thread_checker_.DetachFromThread(); - webrtc::AudioSendStream::Config config(nullptr); - config.voe_channel_id = channel_; - config.rtp.ssrc = ssrc; - stream_ = call_->CreateAudioSendStream(config); - RTC_DCHECK(stream_); + config_.rtp.ssrc = ssrc; + config_.rtp.c_name = c_name; + config_.voe_channel_id = ch; + RecreateAudioSendStream(extensions); } + ~WebRtcAudioSendStream() override { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); Stop(); call_->DestroyAudioSendStream(stream_); } + void RecreateAudioSendStream( + const std::vector<webrtc::RtpExtension>& extensions) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + if (stream_) { + call_->DestroyAudioSendStream(stream_); + stream_ = nullptr; + } + config_.rtp.extensions = extensions; + RTC_DCHECK(!stream_); + stream_ = call_->CreateAudioSendStream(config_); + RTC_CHECK(stream_); + } + + bool SendTelephoneEvent(int payload_type, uint8_t event, + uint32_t duration_ms) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(stream_); + return stream_->SendTelephoneEvent(payload_type, event, duration_ms); + } + + webrtc::AudioSendStream::Stats GetStats() const { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(stream_); + return stream_->GetStats(); + } + // Starts the rendering by setting a sink to the renderer to get data // callback. // This method is called on the libjingle worker thread. // TODO(xians): Make sure Start() is called only once. void Start(AudioRenderer* renderer) { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(renderer); if (renderer_) { RTC_DCHECK(renderer_ == renderer); @@ -1353,16 +1139,11 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream renderer_ = renderer; } - webrtc::AudioSendStream::Stats GetStats() const { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); - return stream_->GetStats(); - } - // Stops rendering by setting the sink of the renderer to nullptr. No data // callback will be received after this method. // This method is called on the libjingle worker thread. void Stop() { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (renderer_) { renderer_->SetSink(nullptr); renderer_ = nullptr; @@ -1374,11 +1155,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream void OnData(const void* audio_data, int bits_per_sample, int sample_rate, - int number_of_channels, + size_t number_of_channels, size_t number_of_frames) override { + RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread()); RTC_DCHECK(voe_audio_transport_); - voe_audio_transport_->OnData(channel_, + voe_audio_transport_->OnData(config_.voe_channel_id, audio_data, bits_per_sample, sample_rate, @@ -1389,7 +1171,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream // Callback from the |renderer_| when it is going away. In case Start() has // never been called, this callback won't be triggered. void OnClose() override { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // Set |renderer_| to nullptr to make sure no more callback will get into // the renderer. renderer_ = nullptr; @@ -1397,16 +1179,18 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream // Accessor to the VoE channel ID. int channel() const { - RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); - return channel_; + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + return config_.voe_channel_id; } private: - rtc::ThreadChecker signal_thread_checker_; + rtc::ThreadChecker worker_thread_checker_; rtc::ThreadChecker audio_capture_thread_checker_; - const int channel_ = -1; webrtc::AudioTransport* const voe_audio_transport_ = nullptr; webrtc::Call* call_ = nullptr; + webrtc::AudioSendStream::Config config_; + // The stream is owned by WebRtcAudioSendStream and may be reallocated if + // configuration changes. webrtc::AudioSendStream* stream_ = nullptr; // Raw pointer to AudioRenderer owned by LocalAudioTrackHandler. @@ -1419,80 +1203,163 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { public: - explicit WebRtcAudioReceiveStream(int voe_channel_id) - : channel_(voe_channel_id) {} + WebRtcAudioReceiveStream(int ch, uint32_t remote_ssrc, uint32_t local_ssrc, + bool use_combined_bwe, const std::string& sync_group, + const std::vector<webrtc::RtpExtension>& extensions, + webrtc::Call* call) + : call_(call), + config_() { + RTC_DCHECK_GE(ch, 0); + RTC_DCHECK(call); + config_.rtp.remote_ssrc = remote_ssrc; + config_.rtp.local_ssrc = local_ssrc; + config_.voe_channel_id = ch; + config_.sync_group = sync_group; + RecreateAudioReceiveStream(use_combined_bwe, extensions); + } - int channel() { return channel_; } + ~WebRtcAudioReceiveStream() { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + call_->DestroyAudioReceiveStream(stream_); + } + + void RecreateAudioReceiveStream( + const std::vector<webrtc::RtpExtension>& extensions) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RecreateAudioReceiveStream(config_.combined_audio_video_bwe, extensions); + } + void RecreateAudioReceiveStream(bool use_combined_bwe) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RecreateAudioReceiveStream(use_combined_bwe, config_.rtp.extensions); + } + + webrtc::AudioReceiveStream::Stats GetStats() const { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + RTC_DCHECK(stream_); + return stream_->GetStats(); + } + + int channel() const { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + return config_.voe_channel_id; + } + + void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + stream_->SetSink(std::move(sink)); + } private: - int channel_; + void RecreateAudioReceiveStream(bool use_combined_bwe, + const std::vector<webrtc::RtpExtension>& extensions) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + if (stream_) { + call_->DestroyAudioReceiveStream(stream_); + stream_ = nullptr; + } + config_.rtp.extensions = extensions; + config_.combined_audio_video_bwe = use_combined_bwe; + RTC_DCHECK(!stream_); + stream_ = call_->CreateAudioReceiveStream(config_); + RTC_CHECK(stream_); + } + + 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; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream); }; -// WebRtcVoiceMediaChannel WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, const AudioOptions& options, webrtc::Call* call) - : engine_(engine), - send_bitrate_setting_(false), - send_bitrate_bps_(0), - options_(), - dtmf_allowed_(false), - desired_playout_(false), - nack_enabled_(false), - playout_(false), - typing_noise_detected_(false), - desired_send_(SEND_NOTHING), - send_(SEND_NOTHING), - call_(call) { + : engine_(engine), call_(call) { LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel"; - RTC_DCHECK(nullptr != call); + RTC_DCHECK(call); engine->RegisterChannel(this); SetOptions(options); } WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel"; - - // Remove any remaining send streams. + // 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); } - - // Remove any remaining receive streams. - while (!receive_channels_.empty()) { - RemoveRecvStream(receive_channels_.begin()->first); + while (!recv_streams_.empty()) { + RemoveRecvStream(recv_streams_.begin()->first); } - RTC_DCHECK(receive_streams_.empty()); - - // Unregister ourselves from the engine. engine()->UnregisterChannel(this); } bool WebRtcVoiceMediaChannel::SetSendParameters( const AudioSendParameters& params) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: " + << params.ToString(); // TODO(pthatcher): Refactor this to be more clean now that we have // all the information at once. - return (SetSendCodecs(params.codecs) && - SetSendRtpHeaderExtensions(params.extensions) && - SetMaxSendBandwidth(params.max_bandwidth_bps) && - SetOptions(params.options)); + + 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->RecreateAudioSendStream(send_rtp_extensions_); + } + } + + if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) { + return false; + } + return SetOptions(params.options); } bool WebRtcVoiceMediaChannel::SetRecvParameters( const AudioRecvParameters& params) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: " + << params.ToString(); // TODO(pthatcher): Refactor this to be more clean now that we have // all the information at once. - return (SetRecvCodecs(params.codecs) && - SetRecvRtpHeaderExtensions(params.extensions)); + + 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->RecreateAudioReceiveStream(recv_rtp_extensions_); + } + } + + return true; } bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "Setting voice channel options: " << options.ToString(); @@ -1503,26 +1370,27 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { // on top. This means there is no way to "clear" options such that // they go back to the engine default. options_.SetAll(options); - - if (send_ != SEND_NOTHING) { - if (!engine()->ApplyOptions(options_)) { - LOG(LS_WARNING) << - "Failed to apply engine options during channel SetOptions."; - return false; - } + if (!engine()->ApplyOptions(options_)) { + LOG(LS_WARNING) << + "Failed to apply engine options during channel SetOptions."; + return false; } if (dscp_option_changed) { rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT; - if (options_.dscp.GetWithDefaultIfUnset(false)) + if (options_.dscp.value_or(false)) { dscp = kAudioDscpValue; + } if (MediaChannel::SetDscp(dscp) != 0) { LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel"; } } // TODO(solenberg): Don't recreate unless options changed. - RecreateAudioReceiveStreams(); + for (auto& it : recv_streams_) { + it.second->RecreateAudioReceiveStream( + options_.combined_audio_video_bwe.value_or(false)); + } LOG(LS_INFO) << "Set voice channel options. Current options: " << options_.ToString(); @@ -1531,7 +1399,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { bool WebRtcVoiceMediaChannel::SetRecvCodecs( const std::vector<AudioCodec>& codecs) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // Set the payload types to be used for incoming media. LOG(LS_INFO) << "Setting receive voice codecs."; @@ -1568,7 +1436,26 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs( PausePlayout(); } - bool result = SetRecvCodecsInternal(new_codecs); + bool result = true; + for (const AudioCodec& codec : new_codecs) { + webrtc::CodecInst voe_codec; + if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) { + LOG(LS_INFO) << ToString(codec); + voe_codec.pltype = codec.id; + for (const auto& ch : recv_streams_) { + if (engine()->voe()->codec()->SetRecPayloadType( + ch.second->channel(), voe_codec) == -1) { + LOG_RTCERR2(SetRecPayloadType, ch.second->channel(), + ToString(voe_codec)); + result = false; + } + } + } else { + LOG(LS_WARNING) << "Unknown codec " << ToString(codec); + result = false; + break; + } + } if (result) { recv_codecs_ = codecs; } @@ -1588,7 +1475,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( engine()->voe()->codec()->SetFECStatus(channel, false); // Scan through the list to figure out the codec to use for sending, along - // with the proper configuration for VAD and DTMF. + // with the proper configuration for VAD. bool found_send_codec = false; webrtc::CodecInst send_codec; memset(&send_codec, 0, sizeof(send_codec)); @@ -1603,7 +1490,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // Ignore codecs we don't know about. The negotiation step should prevent // this, but double-check to be sure. webrtc::CodecInst voe_codec; - if (!engine()->FindWebRtcCodec(codec, &voe_codec)) { + if (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) { LOG(LS_WARNING) << "Unknown codec " << ToString(codec); continue; } @@ -1644,7 +1531,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // Set packet size if the AudioCodec param kCodecParamPTime is set. int ptime_ms = 0; if (codec.GetParam(kCodecParamPTime, &ptime_ms)) { - if (!SetPTimeAsPacketSize(&send_codec, ptime_ms)) { + if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(&send_codec, ptime_ms)) { LOG(LS_WARNING) << "Failed to set packet size for codec " << send_codec.plname; return false; @@ -1687,7 +1574,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // Set Opus internal DTX. LOG(LS_INFO) << "Attempt to " - << GetEnableString(enable_opus_dtx) + << (enable_opus_dtx ? "enable" : "disable") << " Opus DTX on channel " << channel; if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) { @@ -1717,25 +1604,17 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( SetSendBitrateInternal(send_bitrate_bps_); } - // Loop through the codecs list again to config the telephone-event/CN codec. + // Loop through the codecs list again to config the CN codec. for (const AudioCodec& codec : codecs) { // Ignore codecs we don't know about. The negotiation step should prevent // this, but double-check to be sure. webrtc::CodecInst voe_codec; - if (!engine()->FindWebRtcCodec(codec, &voe_codec)) { + if (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) { LOG(LS_WARNING) << "Unknown codec " << ToString(codec); continue; } - // Find the DTMF telephone event "codec" and tell VoiceEngine channels - // about it. - if (IsCodec(codec, kDtmfCodecName)) { - if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType( - channel, codec.id) == -1) { - LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, codec.id); - return false; - } - } else if (IsCodec(codec, kCnCodecName)) { + if (IsCodec(codec, kCnCodecName)) { // Turn voice activity detection/comfort noise on if supported. // Set the wideband CN payload type appropriately. // (narrowband always uses the static payload type 13). @@ -1789,13 +1668,17 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( bool WebRtcVoiceMediaChannel::SetSendCodecs( const std::vector<AudioCodec>& codecs) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + // TODO(solenberg): Validate input - that payload types don't overlap, are + // within range, filter out codecs we don't support, + // redundant codecs etc. - dtmf_allowed_ = false; + // Find the DTMF telephone event "codec" payload type. + dtmf_payload_type_ = rtc::Optional<int>(); for (const AudioCodec& codec : codecs) { - // Find the DTMF telephone event "codec". if (IsCodec(codec, kDtmfCodecName)) { - dtmf_allowed_ = true; + dtmf_payload_type_ = rtc::Optional<int>(codec.id); + break; } } @@ -1808,7 +1691,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( } // Set nack status on receive channels and update |nack_enabled_|. - for (const auto& ch : receive_channels_) { + for (const auto& ch : recv_streams_) { SetNack(ch.second->channel(), nack_enabled_); } @@ -1844,106 +1727,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodec( return true; } -bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions( - const std::vector<RtpHeaderExtension>& extensions) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - if (receive_extensions_ == extensions) { - return true; - } - - for (const auto& ch : receive_channels_) { - if (!SetChannelRecvRtpHeaderExtensions(ch.second->channel(), extensions)) { - return false; - } - } - - receive_extensions_ = extensions; - - // Recreate AudioReceiveStream:s. - { - std::vector<webrtc::RtpExtension> exts; - - const RtpHeaderExtension* audio_level_extension = - FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension); - if (audio_level_extension) { - exts.push_back({ - kRtpAudioLevelHeaderExtension, audio_level_extension->id}); - } - - const RtpHeaderExtension* send_time_extension = - FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); - if (send_time_extension) { - exts.push_back({ - kRtpAbsoluteSenderTimeHeaderExtension, send_time_extension->id}); - } - - recv_rtp_extensions_.swap(exts); - RecreateAudioReceiveStreams(); - } - - return true; -} - -bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions( - int channel_id, const std::vector<RtpHeaderExtension>& extensions) { - const RtpHeaderExtension* audio_level_extension = - FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension); - if (!SetHeaderExtension( - &webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id, - audio_level_extension)) { - return false; - } - - const RtpHeaderExtension* send_time_extension = - FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); - if (!SetHeaderExtension( - &webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id, - send_time_extension)) { - return false; - } - - return true; -} - -bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions( - const std::vector<RtpHeaderExtension>& extensions) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - if (send_extensions_ == extensions) { - return true; - } - - for (const auto& ch : send_streams_) { - if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) { - return false; - } - } - - send_extensions_ = extensions; - return true; -} - -bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions( - int channel_id, const std::vector<RtpHeaderExtension>& extensions) { - const RtpHeaderExtension* audio_level_extension = - FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension); - - if (!SetHeaderExtension( - &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id, - audio_level_extension)) { - return false; - } - - const RtpHeaderExtension* send_time_extension = - FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); - if (!SetHeaderExtension( - &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id, - send_time_extension)) { - return false; - } - - return true; -} - bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) { desired_playout_ = playout; return ChangePlayout(desired_playout_); @@ -1958,12 +1741,12 @@ bool WebRtcVoiceMediaChannel::ResumePlayout() { } bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (playout_ == playout) { return true; } - for (const auto& ch : receive_channels_) { + for (const auto& ch : recv_streams_) { if (!SetPlayout(ch.second->channel(), playout)) { LOG(LS_ERROR) << "SetPlayout " << playout << " on channel " << ch.second->channel() << " failed"; @@ -1995,7 +1778,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) { return true; } - // Apply channel specific options. + // Apply channel specific options when channel is enabled for sending. if (send == SEND_MICROPHONE) { engine()->ApplyOptions(options_); } @@ -2007,13 +1790,6 @@ bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) { } } - // Clear up the options after stopping sending. Since we may previously have - // applied the channel specific options, now apply the original options stored - // in WebRtcVoiceEngine. - if (send == SEND_NOTHING) { - engine()->ApplyOptions(engine()->GetOptions()); - } - send_ = send; return true; } @@ -2039,7 +1815,7 @@ bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, bool enable, const AudioOptions* options, AudioRenderer* renderer) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // TODO(solenberg): The state change should be fully rolled back if any one of // these calls fail. if (!SetLocalRenderer(ssrc, renderer)) { @@ -2068,7 +1844,7 @@ int WebRtcVoiceMediaChannel::CreateVoEChannel() { return id; } -bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) { +bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) { if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) { LOG_RTCERR1(DeRegisterExternalTransport, channel); } @@ -2080,7 +1856,7 @@ bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) { } bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); uint32_t ssrc = sp.first_ssrc(); @@ -2097,33 +1873,12 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { return false; } - // Enable RTCP (for quality stats and feedback messages). - if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) { - LOG_RTCERR2(SetRTCPStatus, channel, 1); - } - - SetChannelSendRtpHeaderExtensions(channel, send_extensions_); - - // Set the local (send) SSRC. - if (engine()->voe()->rtp()->SetLocalSSRC(channel, ssrc) == -1) { - LOG_RTCERR2(SetLocalSSRC, channel, ssrc); - DeleteChannel(channel); - return false; - } - - if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) { - LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname); - DeleteChannel(channel); - return false; - } - // Save the channel to send_streams_, so that RemoveSendStream() can still // delete the channel in case failure happens below. webrtc::AudioTransport* audio_transport = engine()->voe()->base()->audio_transport(); - send_streams_.insert( - std::make_pair(ssrc, - new WebRtcAudioSendStream(channel, audio_transport, ssrc, call_))); + send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream( + channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_))); // Set the current codecs to be used for the new channel. We need to do this // after adding the channel to send_channels_, because of how max bitrate is @@ -2138,10 +1893,10 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { // with the same SSRC in order to send receiver reports. if (send_streams_.size() == 1) { receiver_reports_ssrc_ = ssrc; - for (const auto& ch : receive_channels_) { - int recv_channel = ch.second->channel(); + for (const auto& stream : recv_streams_) { + int recv_channel = stream.second->channel(); if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) { - LOG_RTCERR2(SetLocalSSRC, ch.second->channel(), ssrc); + LOG_RTCERR2(SetLocalSSRC, recv_channel, ssrc); return false; } engine()->voe()->base()->AssociateSendChannel(recv_channel, channel); @@ -2154,7 +1909,9 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { } bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_INFO) << "RemoveSendStream: " << ssrc; + auto it = send_streams_.find(ssrc); if (it == send_streams_.end()) { LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc @@ -2165,15 +1922,12 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { int channel = it->second->channel(); ChangeSend(channel, SEND_NOTHING); - // Delete the WebRtcVoiceChannelRenderer object connected to the channel, - // this will disconnect the audio renderer with the send channel. - delete it->second; - send_streams_.erase(it); - - // Clean up and delete the send channel. + // Clean up and delete the send stream+channel. LOG(LS_INFO) << "Removing audio send stream " << ssrc << " with VoiceEngine channel #" << channel << "."; - if (!DeleteChannel(channel)) { + delete it->second; + send_streams_.erase(it); + if (!DeleteVoEChannel(channel)) { return false; } if (send_streams_.empty()) { @@ -2183,14 +1937,14 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { } bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "AddRecvStream: " << sp.ToString(); if (!ValidateStreamParams(sp)) { return false; } - uint32_t ssrc = sp.first_ssrc(); + const uint32_t ssrc = sp.first_ssrc(); if (ssrc == 0) { LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported."; return false; @@ -2202,114 +1956,87 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { RemoveRecvStream(ssrc); } - if (receive_channels_.find(ssrc) != receive_channels_.end()) { + if (GetReceiveChannelId(ssrc) != -1) { LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc; return false; } - RTC_DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end()); // Create a new channel for receiving audio data. - int channel = CreateVoEChannel(); + const int channel = CreateVoEChannel(); if (channel == -1) { return false; } - if (!ConfigureRecvChannel(channel)) { - DeleteChannel(channel); - return false; - } - - WebRtcAudioReceiveStream* stream = new WebRtcAudioReceiveStream(channel); - receive_channels_.insert(std::make_pair(ssrc, stream)); - receive_stream_params_[ssrc] = sp; - AddAudioReceiveStream(ssrc); - - LOG(LS_INFO) << "New audio stream " << ssrc - << " registered to VoiceEngine channel #" - << channel << "."; - return true; -} - -bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - - int send_channel = GetSendChannelId(receiver_reports_ssrc_); - if (send_channel != -1) { - // Associate receive channel with first send channel (so the receive channel - // can obtain RTT from the send channel) - engine()->voe()->base()->AssociateSendChannel(channel, send_channel); - LOG(LS_INFO) << "VoiceEngine channel #" << channel - << " is associated with channel #" << send_channel << "."; - } - if (engine()->voe()->rtp()->SetLocalSSRC(channel, - receiver_reports_ssrc_) == -1) { - LOG_RTCERR1(SetLocalSSRC, channel); - return false; - } // Turn off all supported codecs. - int ncodecs = engine()->voe()->codec()->NumOfCodecs(); - for (int i = 0; i < ncodecs; ++i) { - webrtc::CodecInst voe_codec; - if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) { - voe_codec.pltype = -1; - if (engine()->voe()->codec()->SetRecPayloadType( - channel, voe_codec) == -1) { - LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec)); - return false; - } + // TODO(solenberg): Remove once "no codecs" is the default state of a stream. + for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) { + voe_codec.pltype = -1; + if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) { + LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec)); + DeleteVoEChannel(channel); + return false; } } // Only enable those configured for this channel. for (const auto& codec : recv_codecs_) { webrtc::CodecInst voe_codec; - if (engine()->FindWebRtcCodec(codec, &voe_codec)) { + if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) { voe_codec.pltype = codec.id; if (engine()->voe()->codec()->SetRecPayloadType( channel, voe_codec) == -1) { LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec)); + DeleteVoEChannel(channel); return false; } } } - SetNack(channel, nack_enabled_); - - // Set RTP header extension for the new channel. - if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) { - return false; + const int send_channel = GetSendChannelId(receiver_reports_ssrc_); + if (send_channel != -1) { + // Associate receive channel with first send channel (so the receive channel + // can obtain RTT from the send channel) + engine()->voe()->base()->AssociateSendChannel(channel, send_channel); + LOG(LS_INFO) << "VoiceEngine channel #" << channel + << " is associated with channel #" << send_channel << "."; } + recv_streams_.insert(std::make_pair(ssrc, new WebRtcAudioReceiveStream( + channel, ssrc, receiver_reports_ssrc_, + options_.combined_audio_video_bwe.value_or(false), sp.sync_label, + recv_rtp_extensions_, call_))); + + SetNack(channel, nack_enabled_); SetPlayout(channel, playout_); + return true; } bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; - auto it = receive_channels_.find(ssrc); - if (it == receive_channels_.end()) { + const auto it = recv_streams_.find(ssrc); + if (it == recv_streams_.end()) { LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc << " which doesn't exist."; return false; } - RemoveAudioReceiveStream(ssrc); - receive_stream_params_.erase(ssrc); - - const int channel = it->second->channel(); - delete it->second; - receive_channels_.erase(it); - // Deregister default channel, if that's the one being destroyed. if (IsDefaultRecvStream(ssrc)) { default_recv_ssrc_ = -1; } - LOG(LS_INFO) << "Removing audio stream " << ssrc + const int channel = it->second->channel(); + + // Clean up and delete the receive stream+channel. + LOG(LS_INFO) << "Removing audio receive stream " << ssrc << " with VoiceEngine channel #" << channel << "."; - return DeleteChannel(channel); + it->second->SetRawAudioSink(nullptr); + delete it->second; + recv_streams_.erase(it); + return DeleteVoEChannel(channel); } bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc, @@ -2337,9 +2064,9 @@ bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc, bool WebRtcVoiceMediaChannel::GetActiveStreams( AudioInfo::StreamList* actives) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); actives->clear(); - for (const auto& ch : receive_channels_) { + for (const auto& ch : recv_streams_) { int level = GetOutputLevel(ch.second->channel()); if (level > 0) { actives->push_back(std::make_pair(ch.first, level)); @@ -2349,9 +2076,9 @@ bool WebRtcVoiceMediaChannel::GetActiveStreams( } int WebRtcVoiceMediaChannel::GetOutputLevel() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); int highest = 0; - for (const auto& ch : receive_channels_) { + for (const auto& ch : recv_streams_) { highest = std::max(GetOutputLevel(ch.second->channel()), highest); } return highest; @@ -2383,7 +2110,7 @@ void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window, } bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); if (ssrc == 0) { default_recv_volume_ = volume; if (default_recv_ssrc_ == -1) { @@ -2408,64 +2135,48 @@ bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { } bool WebRtcVoiceMediaChannel::CanInsertDtmf() { - return dtmf_allowed_; + return dtmf_payload_type_ ? true : false; } -bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, - int event, - int duration, - int flags) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - if (!dtmf_allowed_) { +bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event, + int duration) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf"; + if (!dtmf_payload_type_) { return false; } - // Send the event. - if (flags & cricket::DF_SEND) { - int channel = -1; - if (ssrc == 0) { - if (send_streams_.size() > 0) { - channel = send_streams_.begin()->second->channel(); - } - } else { - channel = GetSendChannelId(ssrc); - } - if (channel == -1) { - LOG(LS_WARNING) << "InsertDtmf - The specified ssrc " - << ssrc << " is not in use."; - return false; - } - // Send DTMF using out-of-band DTMF. ("true", as 3rd arg) - if (engine()->voe()->dtmf()->SendTelephoneEvent( - channel, event, true, duration) == -1) { - LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration); - 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()) { + LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; + return false; } - - // Play the event. - if (flags & cricket::DF_PLAY) { - // Play DTMF tone locally. - if (engine()->voe()->dtmf()->PlayDtmfTone(event, duration) == -1) { - LOG_RTCERR2(PlayDtmfTone, event, duration); - return false; - } + if (event < kMinTelephoneEventCode || + event > kMaxTelephoneEventCode) { + LOG(LS_WARNING) << "DTMF event code " << event << " out of range."; + return false; } - - return true; + if (duration < kMinTelephoneEventDuration || + duration > kMaxTelephoneEventDuration) { + LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range."; + return false; + } + return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration); } void WebRtcVoiceMediaChannel::OnPacketReceived( rtc::Buffer* packet, const rtc::PacketTime& packet_time) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); uint32_t ssrc = 0; if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) { return; } - if (receive_channels_.empty()) { - // Create new channel, which will be the default receive channel. + // If we don't have a default channel, and the SSRC is unknown, create a + // default channel. + if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) { StreamParams sp; sp.ssrcs.push_back(ssrc); LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; @@ -2485,7 +2196,13 @@ void WebRtcVoiceMediaChannel::OnPacketReceived( reinterpret_cast<const uint8_t*>(packet->data()), packet->size(), webrtc_packet_time); if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) { - return; + // If the SSRC is unknown here, route it to the default channel, if we have + // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208 + if (default_recv_ssrc_ == -1) { + return; + } else { + ssrc = default_recv_ssrc_; + } } // Find the channel to send this packet to. It must exist since webrtc::Call @@ -2500,7 +2217,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived( void WebRtcVoiceMediaChannel::OnRtcpReceived( rtc::Buffer* packet, const rtc::PacketTime& packet_time) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); // Forward packet to Call as well. const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, @@ -2542,7 +2259,7 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived( } bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); int channel = GetSendChannelId(ssrc); if (channel == -1) { LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use."; @@ -2601,7 +2318,7 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) { return true; webrtc::CodecInst codec = *send_codec_; - bool is_multi_rate = IsCodecMultiRate(codec); + bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec); if (is_multi_rate) { // If codec is multi-rate then just set the bitrate. @@ -2629,7 +2346,7 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) { } bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); RTC_DCHECK(info); // Get SSRC and stats for each sender. @@ -2652,15 +2369,14 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { sinfo.echo_delay_std_ms = stats.echo_delay_std_ms; sinfo.echo_return_loss = stats.echo_return_loss; sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement; - sinfo.typing_noise_detected = typing_noise_detected_; - // TODO(solenberg): Move to AudioSendStream. - // sinfo.typing_noise_detected = stats.typing_noise_detected; + sinfo.typing_noise_detected = + (send_ == SEND_NOTHING ? false : stats.typing_noise_detected); info->senders.push_back(sinfo); } // Get SSRC and stats for each receiver. RTC_DCHECK(info->receivers.size() == 0); - for (const auto& stream : receive_streams_) { + for (const auto& stream : recv_streams_) { webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats(); VoiceReceiverInfo rinfo; rinfo.add_ssrc(stats.remote_ssrc); @@ -2694,15 +2410,17 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { return true; } -void WebRtcVoiceMediaChannel::OnError(int error) { - if (send_ == SEND_NOTHING) { +void WebRtcVoiceMediaChannel::SetRawAudioSink( + uint32_t ssrc, + rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink"; + const auto it = recv_streams_.find(ssrc); + if (it == recv_streams_.end()) { + LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc; return; } - if (error == VE_TYPING_NOISE_WARNING) { - typing_noise_detected_ = true; - } else if (error == VE_TYPING_NOISE_OFF_WARNING) { - typing_noise_detected_ = false; - } + it->second->SetRawAudioSink(std::move(sink)); } int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) { @@ -2712,16 +2430,16 @@ int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) { } int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - const auto it = receive_channels_.find(ssrc); - if (it != receive_channels_.end()) { + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); + const auto it = recv_streams_.find(ssrc); + if (it != recv_streams_.end()) { return it->second->channel(); } return -1; } int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); const auto it = send_streams_.find(ssrc); if (it != send_streams_.end()) { return it->second->channel(); @@ -2762,7 +2480,7 @@ bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec, if (codec.id == red_pt) { // If we find the right codec, that will be the codec we pass to // SetSendCodec, with the desired payload type. - if (engine()->FindWebRtcCodec(codec, send_codec)) { + if (WebRtcVoiceEngine::ToCodecInst(codec, send_codec)) { return true; } else { break; @@ -2786,117 +2504,6 @@ bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) { } return true; } - -// Convert VoiceEngine error code into VoiceMediaChannel::Error enum. -VoiceMediaChannel::Error - WebRtcVoiceMediaChannel::WebRtcErrorToChannelError(int err_code) { - switch (err_code) { - case 0: - return ERROR_NONE; - case VE_CANNOT_START_RECORDING: - case VE_MIC_VOL_ERROR: - case VE_GET_MIC_VOL_ERROR: - case VE_CANNOT_ACCESS_MIC_VOL: - return ERROR_REC_DEVICE_OPEN_FAILED; - case VE_SATURATION_WARNING: - return ERROR_REC_DEVICE_SATURATION; - case VE_REC_DEVICE_REMOVED: - return ERROR_REC_DEVICE_REMOVED; - case VE_RUNTIME_REC_WARNING: - case VE_RUNTIME_REC_ERROR: - return ERROR_REC_RUNTIME_ERROR; - case VE_CANNOT_START_PLAYOUT: - case VE_SPEAKER_VOL_ERROR: - case VE_GET_SPEAKER_VOL_ERROR: - case VE_CANNOT_ACCESS_SPEAKER_VOL: - return ERROR_PLAY_DEVICE_OPEN_FAILED; - case VE_RUNTIME_PLAY_WARNING: - case VE_RUNTIME_PLAY_ERROR: - return ERROR_PLAY_RUNTIME_ERROR; - case VE_TYPING_NOISE_WARNING: - return ERROR_REC_TYPING_NOISE_DETECTED; - default: - return VoiceMediaChannel::ERROR_OTHER; - } -} - -bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter, - int channel_id, const RtpHeaderExtension* extension) { - bool enable = false; - int id = 0; - std::string uri; - if (extension) { - enable = true; - id = extension->id; - uri = extension->uri; - } - if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) { - LOG_RTCERR4(*setter, uri, channel_id, enable, id); - return false; - } - return true; -} - -void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - for (const auto& it : receive_channels_) { - RemoveAudioReceiveStream(it.first); - } - for (const auto& it : receive_channels_) { - AddAudioReceiveStream(it.first); - } -} - -void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - WebRtcAudioReceiveStream* stream = receive_channels_[ssrc]; - RTC_DCHECK(stream != nullptr); - RTC_DCHECK(receive_streams_.find(ssrc) == receive_streams_.end()); - webrtc::AudioReceiveStream::Config config; - config.rtp.remote_ssrc = ssrc; - // Only add RTP extensions if we support combined A/V BWE. - config.rtp.extensions = recv_rtp_extensions_; - config.combined_audio_video_bwe = - options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false); - config.voe_channel_id = stream->channel(); - config.sync_group = receive_stream_params_[ssrc].sync_label; - webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config); - receive_streams_.insert(std::make_pair(ssrc, s)); -} - -void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32_t ssrc) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - auto stream_it = receive_streams_.find(ssrc); - if (stream_it != receive_streams_.end()) { - call_->DestroyAudioReceiveStream(stream_it->second); - receive_streams_.erase(stream_it); - } -} - -bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal( - const std::vector<AudioCodec>& new_codecs) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - for (const AudioCodec& codec : new_codecs) { - webrtc::CodecInst voe_codec; - if (engine()->FindWebRtcCodec(codec, &voe_codec)) { - LOG(LS_INFO) << ToString(codec); - voe_codec.pltype = codec.id; - for (const auto& ch : receive_channels_) { - if (engine()->voe()->codec()->SetRecPayloadType( - ch.second->channel(), voe_codec) == -1) { - LOG_RTCERR2(SetRecPayloadType, ch.second->channel(), - ToString(voe_codec)); - return false; - } - } - } else { - LOG(LS_WARNING) << "Unknown codec " << ToString(codec); - return false; - } - } - return true; -} - } // namespace cricket #endif // HAVE_WEBRTC_VOICE |