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gfiber / vendor / opensource / libjingle / 7246b07650052c150a237c99284fc689388f321c / . / talk / session / phone / webrtcvoiceengine.cc
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/*
* libjingle
* Copyright 2004--2011, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#ifdef HAVE_WEBRTC_VOICE
#include "talk/session/phone/webrtcvoiceengine.h"
#include <algorithm>
#include <cstdio>
#include <string>
#include <vector>
#include "talk/base/base64.h"
#include "talk/base/byteorder.h"
#include "talk/base/common.h"
#include "talk/base/helpers.h"
#include "talk/base/logging.h"
#include "talk/base/stringencode.h"
#include "talk/base/stringutils.h"
#include "talk/session/phone/voiceprocessor.h"
#include "talk/session/phone/webrtcvoe.h"
#ifdef WIN32
#include <objbase.h> // NOLINT
#endif
namespace cricket {
// 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
static const int kDefaultAudioDeviceId = -1;
static const int kDefaultSoundclipDeviceId = -2;
#else
static const int kDefaultAudioDeviceId = 0;
#endif
// extension header for audio levels, as defined in
// http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
static const char kRtpAudioLevelHeaderExtension[] =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
static const char kIsacCodecName[] = "ISAC";
static const char kL16CodecName[] = "L16";
// Dumps an AudioCodec in RFC 2327-ish format.
static std::string ToString(const AudioCodec& codec) {
std::stringstream ss;
ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
<< " (" << codec.id << ")";
return ss.str();
}
static std::string ToString(const webrtc::CodecInst& codec) {
std::stringstream ss;
ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
<< " (" << codec.pltype << ")";
return ss.str();
}
static void LogMultiline(talk_base::LoggingSeverity sev, char* text) {
const char* delim = "\r\n";
for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
LOG_V(sev) << tok;
}
}
// WebRtcVoiceEngine
const WebRtcVoiceEngine::CodecPref WebRtcVoiceEngine::kCodecPrefs[] = {
{ "ISAC", 16000, 103 },
{ "ISAC", 32000, 104 },
{ "speex", 16000, 107 },
{ "G722", 16000, 9 },
{ "ILBC", 8000, 102 },
{ "speex", 8000, 108 },
{ "PCMU", 8000, 0 },
{ "PCMA", 8000, 8 },
{ "CN", 32000, 106 },
{ "CN", 16000, 105 },
{ "CN", 8000, 13 },
{ "red", 8000, 127 },
{ "telephone-event", 8000, 126 },
};
class WebRtcSoundclipMedia : public SoundclipMedia {
public:
explicit WebRtcSoundclipMedia(WebRtcVoiceEngine *engine)
: engine_(engine), webrtc_channel_(-1) {
engine_->RegisterSoundclip(this);
}
virtual ~WebRtcSoundclipMedia() {
engine_->UnregisterSoundclip(this);
if (webrtc_channel_ != -1) {
// We shouldn't have to call Disable() here. DeleteChannel() should call
// StopPlayout() while deleting the channel. We should fix the bug
// inside WebRTC and remove the Disable() call bellow. This work is
// tracked by bug http://b/issue?id=5382855.
PlaySound(NULL, 0, 0);
Disable();
if (engine_->voe_sc()->base()->DeleteChannel(webrtc_channel_)
== -1) {
LOG_RTCERR1(DeleteChannel, webrtc_channel_);
}
}
}
bool Init() {
webrtc_channel_ = engine_->voe_sc()->base()->CreateChannel();
if (webrtc_channel_ == -1) {
LOG_RTCERR0(CreateChannel);
return false;
}
return true;
}
bool Enable() {
if (engine_->voe_sc()->base()->StartPlayout(webrtc_channel_) == -1) {
LOG_RTCERR1(StartPlayout, webrtc_channel_);
return false;
}
return true;
}
bool Disable() {
if (engine_->voe_sc()->base()->StopPlayout(webrtc_channel_) == -1) {
LOG_RTCERR1(StopPlayout, webrtc_channel_);
return false;
}
return true;
}
virtual bool PlaySound(const char *buf, int len, int flags) {
// Must stop playing the current sound (if any), because we are about to
// modify the stream.
if (engine_->voe_sc()->file()->StopPlayingFileLocally(webrtc_channel_)
== -1) {
LOG_RTCERR1(StopPlayingFileLocally, webrtc_channel_);
return false;
}
if (buf) {
stream_.reset(new WebRtcSoundclipStream(buf, len));
stream_->set_loop((flags & SF_LOOP) != 0);
stream_->Rewind();
// Play it.
if (engine_->voe_sc()->file()->StartPlayingFileLocally(
webrtc_channel_, stream_.get()) == -1) {
LOG_RTCERR2(StartPlayingFileLocally, webrtc_channel_, stream_.get());
LOG(LS_ERROR) << "Unable to start soundclip";
return false;
}
} else {
stream_.reset();
}
return true;
}
int GetLastEngineError() const { return engine_->voe_sc()->error(); }
private:
WebRtcVoiceEngine *engine_;
int webrtc_channel_;
talk_base::scoped_ptr<WebRtcSoundclipStream> stream_;
};
WebRtcVoiceEngine::WebRtcVoiceEngine()
: voe_wrapper_(new VoEWrapper()),
voe_wrapper_sc_(new VoEWrapper()),
tracing_(new VoETraceWrapper()),
adm_(NULL),
adm_sc_(NULL),
log_level_(kDefaultLogSeverity),
is_dumping_aec_(false),
desired_local_monitor_enable_(false) {
Construct();
}
WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
VoEWrapper* voe_wrapper_sc,
VoETraceWrapper* tracing)
: voe_wrapper_(voe_wrapper),
voe_wrapper_sc_(voe_wrapper_sc),
tracing_(tracing),
adm_(NULL),
adm_sc_(NULL),
log_level_(kDefaultLogSeverity),
is_dumping_aec_(false),
desired_local_monitor_enable_(false) {
Construct();
}
void WebRtcVoiceEngine::Construct() {
initialized_ = false;
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
ApplyLogging("");
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();
}
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 (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
// Skip uncompressed formats.
if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
continue;
}
const CodecPref* pref = NULL;
for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
if (_stricmp(kCodecPrefs[j].name, voe_codec.plname) == 0 &&
kCodecPrefs[j].clockrate == voe_codec.plfreq) {
pref = &kCodecPrefs[j];
break;
}
}
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));
LOG(LS_INFO) << ToString(codec);
// For ISAC, use 0 to indicate auto bandwidth in our signaling.
if (_stricmp(codec.name.c_str(), kIsacCodecName) == 0) {
codec.bitrate = 0;
}
codecs_.push_back(codec);
} else {
LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
}
}
}
// Make sure they are in local preference order.
std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
LOG_RTCERR0(DeRegisterVoiceEngineObserver);
}
if (adm_) {
voe_wrapper_.reset();
adm_->Release();
adm_ = NULL;
}
if (adm_sc_) {
voe_wrapper_sc_.reset();
adm_sc_->Release();
adm_sc_ = NULL;
}
// Test to see if the media processor was deregistered properly
ASSERT(SignalRxMediaFrame.is_empty());
ASSERT(SignalTxMediaFrame.is_empty());
tracing_->SetTraceCallback(NULL);
}
bool WebRtcVoiceEngine::Init() {
LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
bool res = InitInternal();
if (res) {
LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
} else {
LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
Terminate();
}
return res;
}
bool WebRtcVoiceEngine::InitInternal() {
// Temporarily turn logging level up for the Init call
int old_level = log_level_;
log_level_ = talk_base::_min(log_level_,
static_cast<int>(talk_base::LS_INFO));
ApplyLogging("");
// Init WebRtc VoiceEngine, enabling AEC logging if specified in SetLogging.
if (voe_wrapper_->base()->Init(adm_) == -1) {
LOG_RTCERR0_EX(Init, voe_wrapper_->error());
return false;
}
// Restore the previous log level and apply the log filter.
log_level_ = old_level;
ApplyLogging(log_filter_);
// Log the VoiceEngine version info
char buffer[1024] = "";
voe_wrapper_->base()->GetVersion(buffer);
LOG(LS_INFO) << "WebRtc VoiceEngine Version:";
LogMultiline(talk_base::LS_INFO, buffer);
// Turn on AEC and AGC by default.
if (!SetOptions(
MediaEngineInterface::ECHO_CANCELLATION |
MediaEngineInterface::AUTO_GAIN_CONTROL)) {
return false;
}
// Save the default AGC configuration settings.
if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
LOG_RTCERR0(GetAGCConfig);
return false;
}
// Print our codec list again for the call diagnostic log
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
for (std::vector<AudioCodec>::const_iterator it = codecs_.begin();
it != codecs_.end(); ++it) {
LOG(LS_INFO) << ToString(*it);
}
#if defined(LINUX) && !defined(HAVE_LIBPULSE)
voe_wrapper_sc_->hw()->SetAudioDeviceLayer(webrtc::kAudioLinuxAlsa);
#endif
// Initialize the VoiceEngine instance that we'll use to play out sound clips.
if (voe_wrapper_sc_->base()->Init(adm_sc_) == -1) {
LOG_RTCERR0_EX(Init, voe_wrapper_sc_->error());
return false;
}
// On Windows, tell it to use the default sound (not communication) devices.
// First check whether there is a valid sound device for playback.
// TODO: Clean this up when we support setting the soundclip device.
#ifdef WIN32
int num_of_devices = 0;
if (voe_wrapper_sc_->hw()->GetNumOfPlayoutDevices(num_of_devices) != -1 &&
num_of_devices > 0) {
if (voe_wrapper_sc_->hw()->SetPlayoutDevice(kDefaultSoundclipDeviceId)
== -1) {
LOG_RTCERR1_EX(SetPlayoutDevice, kDefaultSoundclipDeviceId,
voe_wrapper_sc_->error());
return false;
}
} else {
LOG(LS_WARNING) << "No valid sound playout device found.";
}
#endif
initialized_ = true;
return true;
}
void WebRtcVoiceEngine::Terminate() {
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
initialized_ = false;
if (is_dumping_aec_) {
if (voe_wrapper_->processing()->StopDebugRecording() == -1) {
LOG_RTCERR0(StopDebugRecording);
}
is_dumping_aec_ = false;
}
voe_wrapper_sc_->base()->Terminate();
voe_wrapper_->base()->Terminate();
desired_local_monitor_enable_ = false;
}
int WebRtcVoiceEngine::GetCapabilities() {
return AUDIO_SEND | AUDIO_RECV;
}
VoiceMediaChannel *WebRtcVoiceEngine::CreateChannel() {
WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this);
if (!ch->valid()) {
delete ch;
ch = NULL;
}
return ch;
}
SoundclipMedia *WebRtcVoiceEngine::CreateSoundclip() {
WebRtcSoundclipMedia *soundclip = new WebRtcSoundclipMedia(this);
if (!soundclip->Init() || !soundclip->Enable()) {
delete soundclip;
return NULL;
}
return soundclip;
}
bool WebRtcVoiceEngine::SetOptions(int options) {
// NS and typing detection are always on, if supported.
bool aec = (options & MediaEngineInterface::ECHO_CANCELLATION) ? true : false;
bool agc = (options & MediaEngineInterface::AUTO_GAIN_CONTROL) ? true : false;
#if !defined(IOS) && !defined(ANDROID)
if (voe_wrapper_->processing()->SetEcStatus(aec) == -1) {
LOG_RTCERR1(SetEcStatus, aec);
return false;
}
if (aec) {
if (voe_wrapper_->processing()->SetEcMetricsStatus(true) == -1) {
LOG_RTCERR1(SetEcMetricsStatus, true);
return false;
}
}
if (voe_wrapper_->processing()->SetAgcStatus(agc) == -1) {
LOG_RTCERR1(SetAgcStatus, agc);
return false;
}
if (voe_wrapper_->processing()->SetNsStatus(true) == -1) {
LOG_RTCERR1(SetNsStatus, true);
return false;
}
if (voe_wrapper_->processing()->SetTypingDetectionStatus(true) == -1) {
// In case of error, log the info and continue
LOG_RTCERR1(SetTypingDetectionStatus, true);
}
#else
if (voe_wrapper_->processing()->SetEcStatus(aec, kEcAecm) == -1) {
LOG_RTCERR2(SetEcStatus, aec, kEcAecm);
return false;
}
if (aec) {
// Use speakerphone mode with comfort noise generation for mobile.
if (voe_wrapper_->processing()->SetAecmMode(kAecmSpeakerphone, true) != 0) {
LOG_RTCERR2(SetAecmMode, kAecmSpeakerphone, true);
}
}
// On mobile, GIPS recommends fixed AGC (not adaptive)
if (voe_wrapper_->processing()->SetAgcStatus(agc, kAgcFixedDigital) == -1) {
LOG_RTCERR2(SetAgcStatus, agc, kAgcFixedDigital);
return false;
}
// On mobile, GIPS recommends moderate aggressiveness.
if (voe_wrapper_->processing()->SetNsStatus(true,
kNsModerateSuppression) == -1) {
LOG_RTCERR2(SetNsStatus, ns, kNsModerateSuppression);
return false;
}
// No typing detection support on iOS or Android.
#endif // !IOS && !ANDROID
return true;
}
struct ResumeEntry {
ResumeEntry(WebRtcVoiceMediaChannel *c, bool p, SendFlags s)
: channel(c),
playout(p),
send(s) {
}
WebRtcVoiceMediaChannel *channel;
bool playout;
SendFlags send;
};
// TODO: 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) && !defined(ANDROID)
int in_id = in_device ? talk_base::FromString<int>(in_device->id) :
kDefaultAudioDeviceId;
int out_id = out_device ? talk_base::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
<< ")";
// If we're running the local monitor, we need to stop it first.
bool ret = true;
if (!PauseLocalMonitor()) {
LOG(LS_WARNING) << "Failed to pause local monitor";
ret = false;
}
// Must also pause all audio playback and capture.
for (ChannelList::const_iterator i = channels_.begin();
i != channels_.end(); ++i) {
WebRtcVoiceMediaChannel *channel = *i;
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)) {
ret = false;
}
if (ret) {
if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
LOG_RTCERR2(SetRecordingDevice, in_device->name, in_id);
ret = false;
}
}
// 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;
ret = false;
}
if (ret) {
if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
LOG_RTCERR2(SetPlayoutDevice, out_device->name, out_id);
ret = false;
}
}
// Resume all audio playback and capture.
for (ChannelList::const_iterator i = channels_.begin();
i != channels_.end(); ++i) {
WebRtcVoiceMediaChannel *channel = *i;
if (!channel->ResumePlayout()) {
LOG(LS_WARNING) << "Failed to resume playout";
ret = false;
}
if (!channel->ResumeSend()) {
LOG(LS_WARNING) << "Failed to resume send";
ret = false;
}
}
// Resume local monitor.
if (!ResumeLocalMonitor()) {
LOG(LS_WARNING) << "Failed to resume local monitor";
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
<< ")";
}
return ret;
#else
return true;
#endif // !IOS && !ANDROID
}
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.
#ifdef LINUX
*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
}
bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
unsigned int ulevel;
if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
LOG_RTCERR1(GetSpeakerVolume, level);
return false;
}
*level = ulevel;
return true;
}
bool WebRtcVoiceEngine::SetOutputVolume(int level) {
ASSERT(level >= 0 && level <= 255);
if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
LOG_RTCERR1(SetSpeakerVolume, level);
return false;
}
return true;
}
int WebRtcVoiceEngine::GetInputLevel() {
unsigned int ulevel;
return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
static_cast<int>(ulevel) : -1;
}
bool WebRtcVoiceEngine::SetLocalMonitor(bool enable) {
desired_local_monitor_enable_ = enable;
return ChangeLocalMonitor(desired_local_monitor_enable_);
}
bool WebRtcVoiceEngine::ChangeLocalMonitor(bool enable) {
if (enable && !monitor_.get()) {
monitor_.reset(new WebRtcMonitorStream);
if (voe_wrapper_->file()->StartRecordingMicrophone(monitor_.get()) == -1) {
LOG_RTCERR1(StartRecordingMicrophone, monitor_.get());
// Must call Stop() because there are some cases where Start will report
// failure but still change the state, and if we leave VE in the on state
// then it could crash later when trying to invoke methods on our monitor.
voe_wrapper_->file()->StopRecordingMicrophone();
monitor_.reset();
return false;
}
} else if (!enable && monitor_.get()) {
voe_wrapper_->file()->StopRecordingMicrophone();
monitor_.reset();
}
return true;
}
bool WebRtcVoiceEngine::PauseLocalMonitor() {
return ChangeLocalMonitor(false);
}
bool WebRtcVoiceEngine::ResumeLocalMonitor() {
return ChangeLocalMonitor(desired_local_monitor_enable_);
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
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 (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
// Allow arbitrary rates for ISAC to be specified.
if (_stricmp(codec.name.c_str(), kIsacCodecName) == 0) {
codec.bitrate = 0;
}
if (codec.Matches(in)) {
if (out) {
// Fixup the payload type.
voe_codec.pltype = in.id;
// If ISAC is being used, and an explicit bitrate is not specified,
// enable auto bandwidth adjustment.
if (_stricmp(codec.name.c_str(), kIsacCodecName) == 0) {
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
}
return true;
}
}
}
return false;
}
void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) {
// if min_sev == -1, we keep the current log level.
if (min_sev >= 0) {
log_level_ = min_sev;
}
log_filter_ = filter;
ApplyLogging(initialized_ ? log_filter_ : "");
}
int WebRtcVoiceEngine::GetLastEngineError() {
return voe_wrapper_->error();
}
// 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::ApplyLogging(const std::string& log_filter) {
// Set log level.
int filter = 0;
switch (log_level_) {
case talk_base::LS_VERBOSE:
filter |= webrtc::kTraceAll; // fall through
case talk_base::LS_INFO:
filter |= webrtc::kTraceStateInfo; // fall through
case talk_base::LS_WARNING:
filter |= (webrtc::kTraceInfo | webrtc::kTraceWarning); // fall through
case talk_base::LS_ERROR:
filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
}
tracing_->SetTraceFilter(filter);
// Set encrypted trace file.
std::vector<std::string> opts;
talk_base::tokenize(log_filter, ' ', '"', '"', &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);
}
}
// 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() && !is_dumping_aec_) {
// Start dumping AEC when we are not dumping and recordEC has a filename.
if (voe_wrapper_->processing()->StartDebugRecording(
recordEC->c_str()) == -1) {
LOG_RTCERR0(StartDebugRecording);
} else {
is_dumping_aec_ = true;
}
} else if (recordEC == opts.end() && is_dumping_aec_) {
// Stop dumping EC when we are dumping and recordEC has no filename.
if (voe_wrapper_->processing()->StopDebugRecording() == -1) {
LOG_RTCERR0(StopDebugRecording);
}
is_dumping_aec_ = false;
}
}
}
// Ignore spammy trace messages, mostly from the stats API when we haven't
// gotten RTCP info yet from the remote side.
bool WebRtcVoiceEngine::ShouldIgnoreTrace(const std::string& trace) {
static const char* kTracesToIgnore[] = {
"\tfailed to GetReportBlockInformation",
"GetRecCodec() failed to get received codec",
"GetReceivedRtcpStatistics: Could not get received RTP statistics",
"GetRemoteRTCPData() failed to measure statistics due to lack of received RTP and/or RTCP packets", // NOLINT
"GetRemoteRTCPData() failed to retrieve sender info for remote side",
"GetRTPStatistics() failed to measure RTT since no RTP packets have been received yet", // NOLINT
"GetRTPStatistics() failed to read RTP statistics from the RTP/RTCP module",
"GetRTPStatistics() failed to retrieve RTT from the RTP/RTCP module",
"SenderInfoReceived No received SR",
"StatisticsRTP() no statisitics availble",
NULL
};
for (const char* const* p = kTracesToIgnore; *p; ++p) {
if (trace.find(*p) == 0) {
return true;
}
}
return false;
}
void WebRtcVoiceEngine::Print(const webrtc::TraceLevel level,
const char* trace, const int length) {
talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
sev = talk_base::LS_ERROR;
else if (level == webrtc::kTraceWarning)
sev = talk_base::LS_WARNING;
else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
sev = talk_base::LS_INFO;
if (sev >= log_level_) {
// Skip past boilerplate prefix text
if (length < 72) {
std::string msg(trace, length);
LOG(LS_ERROR) << "Malformed webrtc log message: ";
LOG_V(sev) << msg;
} else {
std::string msg(trace + 71, length - 72);
if (!ShouldIgnoreTrace(msg)) {
LOG_V(sev) << "WebRtc:" << msg;
}
}
}
}
void WebRtcVoiceEngine::CallbackOnError(const int channel_num,
const int err_code) {
talk_base::CritScope lock(&channels_cs_);
WebRtcVoiceMediaChannel* channel = NULL;
uint32 ssrc = 0;
LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
<< channel_num << ".";
if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
ASSERT(channel != NULL);
channel->OnError(ssrc, err_code);
} else {
LOG(LS_ERROR) << "VoiceEngine channel " << channel_num
<< " could not be found in channel list when error reported.";
}
}
bool WebRtcVoiceEngine::FindChannelAndSsrc(
int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
ASSERT(channel != NULL && ssrc != NULL);
*channel = NULL;
*ssrc = 0;
// Find corresponding channel and ssrc
for (ChannelList::const_iterator it = channels_.begin();
it != channels_.end(); ++it) {
ASSERT(*it != NULL);
if ((*it)->FindSsrc(channel_num, ssrc)) {
*channel = *it;
return true;
}
}
return false;
}
// This method will search through the WebRtcVoiceMediaChannels and
// obtain the voice engine's channel number.
bool WebRtcVoiceEngine::FindChannelNumFromSsrc(
uint32 ssrc, MediaProcessorDirection direction, int* channel_num) {
ASSERT(channel_num != NULL);
*channel_num = -1;
// Find corresponding channel for ssrc.
for (ChannelList::const_iterator it = channels_.begin();
it != channels_.end(); ++it) {
ASSERT(*it != NULL);
uint32 local_ssrc;
if (voe()->rtp()->GetLocalSSRC((*it)->voe_channel(), local_ssrc) != -1) {
if (ssrc == local_ssrc) {
*channel_num = (*it)->voe_channel();
}
}
if (*channel_num == -1 && (direction & MPD_RX) != 0) {
*channel_num = (*it)->GetChannelNum(ssrc);
}
if (*channel_num != -1) {
return true;
}
}
LOG(LS_WARNING) << "FindChannelFromSsrc. No Channel Found for Ssrc: " << ssrc;
return false;
}
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel *channel) {
talk_base::CritScope lock(&channels_cs_);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
talk_base::CritScope lock(&channels_cs_);
ChannelList::iterator i = std::find(channels_.begin(),
channels_.end(),
channel);
if (i != channels_.end()) {
channels_.erase(i);
}
}
void WebRtcVoiceEngine::RegisterSoundclip(WebRtcSoundclipMedia *soundclip) {
soundclips_.push_back(soundclip);
}
void WebRtcVoiceEngine::UnregisterSoundclip(WebRtcSoundclipMedia *soundclip) {
SoundclipList::iterator i = std::find(soundclips_.begin(),
soundclips_.end(),
soundclip);
if (i != soundclips_.end()) {
soundclips_.erase(i);
}
}
// 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) {
webrtc::AgcConfig config = default_agc_config_;
config.targetLeveldBOv += delta;
LOG(LS_INFO) << "Adjusting AGC level from default -"
<< default_agc_config_.targetLeveldBOv << "dB to -"
<< config.targetLeveldBOv << "dB";
if (voe_wrapper_->processing()->SetAgcConfig(config) == -1) {
LOG_RTCERR1(SetAgcConfig, config.targetLeveldBOv);
return false;
}
return true;
}
// Configures echo cancellation and noise suppression modes according to
// whether or not we are in a multi-point conference.
bool WebRtcVoiceEngine::SetConferenceMode(bool enable) {
// Only use EC_AECM for mobile.
#if defined(IOS) || defined(ANDROID)
return true;
#endif
LOG(LS_INFO) << (enable ? "Enabling" : "Disabling")
<< " Conference Mode noise reduction";
// We always configure noise suppression on, so just toggle the mode.
const webrtc::NsModes ns_mode = enable ? webrtc::kNsConference
: webrtc::kNsDefault;
if (voe_wrapper_->processing()->SetNsStatus(true, ns_mode) == -1) {
LOG_RTCERR2(SetNsStatus, true, ns_mode);
return false;
}
// Echo-cancellation is a user-option, so preserve the enable state and
// just toggle the mode.
bool aec;
webrtc::EcModes ec_mode;
if (voe_wrapper_->processing()->GetEcStatus(aec, ec_mode) == -1) {
LOG_RTCERR0(GetEcStatus);
return false;
}
ec_mode = enable ? webrtc::kEcConference : webrtc::kEcDefault;
if (voe_wrapper_->processing()->SetEcStatus(aec, ec_mode) == -1) {
LOG_RTCERR2(SetEcStatus, aec, ec_mode);
return false;
}
return true;
}
bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc) {
if (initialized_) {
LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
return false;
}
if (adm_) {
adm_->Release();
adm_ = NULL;
}
if (adm) {
adm_ = adm;
adm_->AddRef();
}
if (adm_sc_) {
adm_sc_->Release();
adm_sc_ = NULL;
}
if (adm_sc) {
adm_sc_ = adm_sc;
adm_sc_->AddRef();
}
return true;
}
bool WebRtcVoiceEngine::RegisterProcessor(
uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection direction) {
bool register_with_webrtc = false;
int channel_id = -1;
bool success = false;
bool found_channel = FindChannelNumFromSsrc(ssrc, direction, &channel_id);
if (voice_processor == NULL || !found_channel) {
LOG(LS_WARNING) << "Media Processing Registration Failed. ssrc: " << ssrc
<< " foundChannel: " << found_channel;
return false;
}
talk_base::CritScope cs(&signal_media_critical_);
webrtc::ProcessingTypes processing_type;
if (direction == MPD_RX) {
processing_type = webrtc::kPlaybackAllChannelsMixed;
if (SignalRxMediaFrame.is_empty()) {
register_with_webrtc = true;
}
SignalRxMediaFrame.connect(voice_processor,
&VoiceProcessor::OnFrame);
} else {
processing_type = webrtc::kRecordingPerChannel;
if (SignalTxMediaFrame.is_empty()) {
register_with_webrtc = true;
}
SignalTxMediaFrame.connect(voice_processor,
&VoiceProcessor::OnFrame);
}
if (register_with_webrtc) {
if (voe()->media()->
RegisterExternalMediaProcessing(channel_id,
processing_type,
*this) == -1) {
LOG_RTCERR2(RegisterExternalMediaProcessing,
channel_id,
processing_type);
success = false;
} else {
LOG(LS_INFO) << "Media Processing Registration Succeeded. channel:"
<< channel_id;
success = true;
}
} else {
// If we don't have to register with the engine, we just needed to
// connect a new processor, set success to true;
success = true;
}
return success;
}
bool WebRtcVoiceEngine::UnregisterProcessor(
uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection direction) {
int channel_id = -1;
bool found_channel = FindChannelNumFromSsrc(ssrc, direction, &channel_id);
bool success = true;
if (voice_processor == NULL || !found_channel) {
LOG(LS_WARNING) << "Media Processing Deregistration Failed. ssrc: "
<< ssrc
<< " foundChannel: "
<< found_channel;
return false;
}
talk_base::CritScope cs(&signal_media_critical_);
if ((direction & MPD_RX) != 0) {
SignalRxMediaFrame.disconnect(voice_processor);
if (SignalRxMediaFrame.is_empty()) {
if (voe()->media()->DeRegisterExternalMediaProcessing(channel_id,
webrtc::kPlaybackAllChannelsMixed) != -1) {
LOG(LS_INFO) << "Media Processing DeRegistration Succeeded. channel:"
<< channel_id;
} else {
LOG_RTCERR2(DeRegisterExternalMediaProcessing,
channel_id,
webrtc::kPlaybackAllChannelsMixed);
success = false;
}
}
}
if ((direction & MPD_TX) != 0) {
SignalTxMediaFrame.disconnect(voice_processor);
if (SignalTxMediaFrame.is_empty()) {
if (voe()->media()->DeRegisterExternalMediaProcessing(channel_id,
webrtc::kRecordingPerChannel) != -1) {
LOG(LS_INFO) << "Media Processing DeRegistration Succeeded. channel:"
<< channel_id;
} else {
LOG_RTCERR2(DeRegisterExternalMediaProcessing,
channel_id,
webrtc::kRecordingPerChannel);
success = false;
}
}
}
return success;
}
// Implementing method from WebRtc VoEMediaProcess interface
void WebRtcVoiceEngine::Process(const int channel,
const webrtc::ProcessingTypes type,
WebRtc_Word16 audio10ms[],
const int length,
const int sampling_freq,
const bool is_stereo) {
uint32 ssrc;
WebRtcVoiceMediaChannel* media_channel;
if (FindChannelAndSsrc(channel, &media_channel, &ssrc)) {
talk_base::CritScope cs(&signal_media_critical_);
AudioFrame frame(audio10ms, length, sampling_freq, is_stereo);
if (type == webrtc::kPlaybackAllChannelsMixed) {
SignalRxMediaFrame(ssrc, &frame);
} else if (type == webrtc::kRecordingPerChannel) {
SignalTxMediaFrame(ssrc, &frame);
}
} else {
LOG(LS_WARNING) << "MediaProcess Callback invoked with unexpected channel: "
<< channel;
}
}
// WebRtcVoiceMediaChannel
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->voe()->base()->CreateChannel()),
channel_options_(0),
agc_adjusted_(false),
dtmf_allowed_(false),
desired_playout_(false),
playout_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
<< voe_channel();
// Register external transport
if (engine->voe()->network()->RegisterExternalTransport(
voe_channel(), *static_cast<Transport*>(this)) == -1) {
LOG_RTCERR2(RegisterExternalTransport, voe_channel(), this);
}
// Enable RTCP (for quality stats and feedback messages)
EnableRtcp(voe_channel());
// Create a random but nonzero send SSRC
SetSendSsrc(talk_base::CreateRandomNonZeroId());
// Reset all recv codecs; they will be enabled via SetRecvCodecs.
ResetRecvCodecs(voe_channel());
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
// DeRegister external transport
if (engine()->voe()->network()->DeRegisterExternalTransport(
voe_channel()) == -1) {
LOG_RTCERR1(DeRegisterExternalTransport, voe_channel());
}
// Unregister ourselves from the engine.
engine()->UnregisterChannel(this);
// Remove any remaining streams.
while (!mux_channels_.empty()) {
RemoveStream(mux_channels_.begin()->first);
}
// Delete the primary channel.
if (engine()->voe()->base()->DeleteChannel(voe_channel()) == -1) {
LOG_RTCERR1(DeleteChannel, voe_channel());
}
}
bool WebRtcVoiceMediaChannel::SetOptions(int flags) {
// Always accept flags that are unchanged.
if (channel_options_ == flags) {
return true;
}
// Reject new options if we're already sending.
if (send_ != SEND_NOTHING) {
return false;
}
// Save the options, to be interpreted where appropriate.
channel_options_ = flags;
return true;
}
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
// Set the payload types to be used for incoming media.
bool ret = true;
LOG(LS_INFO) << "Setting receive voice codecs:";
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end() && ret; ++it) {
webrtc::CodecInst voe_codec;
if (engine()->FindWebRtcCodec(*it, &voe_codec)) {
LOG(LS_INFO) << ToString(*it);
voe_codec.pltype = it->id;
if (engine()->voe()->codec()->SetRecPayloadType(
voe_channel(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, voe_channel(), ToString(voe_codec));
ret = false;
}
} else {
LOG(LS_WARNING) << "Unknown codec " << ToString(*it);
ret = false;
}
}
return ret;
}
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
// Disable DTMF, VAD, and FEC unless we know the other side wants them.
dtmf_allowed_ = false;
engine()->voe()->codec()->SetVADStatus(voe_channel(), false);
engine()->voe()->rtp()->SetFECStatus(voe_channel(), false);
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD and DTMF.
bool first = true;
webrtc::CodecInst send_codec;
memset(&send_codec, 0, sizeof(send_codec));
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
// 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(*it, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(voe_codec);
continue;
}
// Find the DTMF telephone event "codec" and tell VoiceEngine about it.
if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
_stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
voe_channel(), it->id);
dtmf_allowed_ = true;
}
// Turn voice activity detection/comfort noise on if supported.
// Set the wideband CN payload type appropriately.
// (narrowband always uses the static payload type 13).
if (_stricmp(it->name.c_str(), "CN") == 0) {
webrtc::PayloadFrequencies cn_freq;
switch (it->clockrate) {
case 8000:
cn_freq = webrtc::kFreq8000Hz;
break;
case 16000:
cn_freq = webrtc::kFreq16000Hz;
break;
case 32000:
cn_freq = webrtc::kFreq32000Hz;
break;
default:
LOG(LS_WARNING) << "CN frequency " << it->clockrate
<< " not supported.";
continue;
}
engine()->voe()->codec()->SetVADStatus(voe_channel(), true);
if (cn_freq != webrtc::kFreq8000Hz) {
engine()->voe()->codec()->SetSendCNPayloadType(voe_channel(),
it->id, cn_freq);
}
}
// We'll use the first codec in the list to actually send audio data.
// Be sure to use the payload type requested by the remote side.
// "red", for FEC audio, is a special case where the actual codec to be
// used is specified in params.
if (first) {
if (_stricmp(it->name.c_str(), "red") == 0) {
// Parse out the RED parameters. If we fail, just ignore RED;
// we don't support all possible params/usage scenarios.
if (!GetRedSendCodec(*it, codecs, &send_codec)) {
continue;
}
// Enable redundant encoding of the specified codec. Treat any
// failure as a fatal internal error.
LOG(LS_INFO) << "Enabling FEC";
if (engine()->voe()->rtp()->SetFECStatus(voe_channel(),
true, it->id) == -1) {
LOG_RTCERR3(SetFECStatus, voe_channel(), true, it->id);
return false;
}
} else {
send_codec = voe_codec;
}
first = false;
}
}
// If we're being asked to set an empty list of codecs, due to a buggy client,
// choose the most common format: PCMU
if (first) {
LOG(LS_WARNING) << "Received empty list of codecs; using PCMU/8000";
AudioCodec codec(0, "PCMU", 8000, 0, 1, 0);
engine()->FindWebRtcCodec(codec, &send_codec);
}
// Set the codec.
LOG(LS_INFO) << "Selected voice codec " << ToString(send_codec)
<< ", bitrate=" << send_codec.rate;
if (engine()->voe()->codec()->SetSendCodec(voe_channel(),
send_codec) == -1) {
LOG_RTCERR2(SetSendCodec, voe_channel(), ToString(send_codec));
return false;
}
send_codec_.reset(new webrtc::CodecInst(send_codec));
return true;
}
bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
// We don't support any incoming extensions headers right now.
return true;
}
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
// Enable the audio level extension header if requested.
std::vector<RtpHeaderExtension>::const_iterator it;
for (it = extensions.begin(); it != extensions.end(); ++it) {
if (it->uri == kRtpAudioLevelHeaderExtension) {
break;
}
}
bool enable = (it != extensions.end());
int id = 0;
if (enable) {
id = it->id;
if (id < kMinRtpHeaderExtensionId ||
id > kMaxRtpHeaderExtensionId) {
LOG(LS_WARNING) << "Invalid RTP header extension id " << id;
return false;
}
}
// This api call is not available in iOS version of VoiceEngine currently.
#if !defined(IOS) && !defined(ANDROID)
if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus(
voe_channel(), enable, id) == -1) {
LOG_RTCERR3(SetRTPAudioLevelIndicationStatus, voe_channel(), enable, id);
return false;
}
#endif
return true;
}
bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
desired_playout_ = playout;
return ChangePlayout(desired_playout_);
}
bool WebRtcVoiceMediaChannel::PausePlayout() {
return ChangePlayout(false);
}
bool WebRtcVoiceMediaChannel::ResumePlayout() {
return ChangePlayout(desired_playout_);
}
bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
if (playout_ == playout) {
return true;
}
bool result = true;
if (mux_channels_.empty()) {
// Only toggle the default channel if we don't have any other channels.
result = SetPlayout(voe_channel(), playout);
}
for (ChannelMap::iterator it = mux_channels_.begin();
it != mux_channels_.end() && result; ++it) {
if (!SetPlayout(it->second, playout)) {
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel " << it->second
<< " failed";
result = false;
}
}
if (result) {
playout_ = playout;
}
return result;
}
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
desired_send_ = send;
return ChangeSend(desired_send_);
}
bool WebRtcVoiceMediaChannel::PauseSend() {
return ChangeSend(SEND_NOTHING);
}
bool WebRtcVoiceMediaChannel::ResumeSend() {
return ChangeSend(desired_send_);
}
bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
if (send_ == send) {
return true;
}
if (send == SEND_MICROPHONE) {
#ifdef CHROMEOS
// Conference mode doesn't work well on ChromeOS.
if (!engine()->SetConferenceMode(false)) {
LOG_RTCERR1(SetConferenceMode, voe_channel());
return false;
}
#else
// Multi-point conferences use conference-mode noise filtering.
if (!engine()->SetConferenceMode(
0 != (channel_options_ & OPT_CONFERENCE))) {
LOG_RTCERR1(SetConferenceMode, voe_channel());
return false;
}
#endif // CHROMEOS
if ((channel_options_ & OPT_AGC_MINUS_10DB) && !agc_adjusted_) {
if (engine()->AdjustAgcLevel(kMinus10DbAdjustment)) {
agc_adjusted_ = true;
}
}
// VoiceEngine resets sequence number when StopSend is called. This
// sometimes causes libSRTP to complain about packets being
// replayed. To get around this we store the last sent sequence
// number and initializes the channel with the next to continue on
// the same sequence.
if (sequence_number() != -1) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel restores seqnum="
<< sequence_number() + 1;
if (engine()->voe()->sync()->SetInitSequenceNumber(
voe_channel(), sequence_number() + 1) == -1) {
LOG_RTCERR2(SetInitSequenceNumber, voe_channel(),
sequence_number() + 1);
}
}
if (engine()->voe()->base()->StartSend(voe_channel()) == -1) {
LOG_RTCERR1(StartSend, voe_channel());
return false;
}
if (engine()->voe()->file()->StopPlayingFileAsMicrophone(
voe_channel()) == -1) {
LOG_RTCERR1(StopPlayingFileAsMicrophone, voe_channel());
return false;
}
} else if (send == SEND_RINGBACKTONE) {
ASSERT(ringback_tone_.get() != NULL);
if (!ringback_tone_.get()) {
return false;
}
if (engine()->voe()->file()->StartPlayingFileAsMicrophone(
voe_channel(), ringback_tone_.get(), false) == -1) {
LOG_RTCERR3(StartPlayingFileAsMicrophone, voe_channel(),
ringback_tone_.get(), false);
return false;
}
// VoiceEngine resets sequence number when StopSend is called. This
// sometimes causes libSRTP to complain about packets being
// replayed. To get around this we store the last sent sequence
// number and initializes the channel with the next to continue on
// the same sequence.
if (sequence_number() != -1) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel restores seqnum="
<< sequence_number() + 1;
if (engine()->voe()->sync()->SetInitSequenceNumber(
voe_channel(), sequence_number() + 1) == -1) {
LOG_RTCERR2(SetInitSequenceNumber, voe_channel(),
sequence_number() + 1);
}
}
if (engine()->voe()->base()->StartSend(voe_channel()) == -1) {
LOG_RTCERR1(StartSend, voe_channel());
return false;
}
} else { // SEND_NOTHING
if (engine()->voe()->base()->StopSend(voe_channel()) == -1) {
LOG_RTCERR1(StopSend, voe_channel());
}
// Reset the AGC level, if it was set.
if (agc_adjusted_) {
if (engine()->AdjustAgcLevel(0)) {
agc_adjusted_ = false;
}
}
// Disable conference-mode noise filtering.
if (!engine()->SetConferenceMode(false)) {
LOG_RTCERR1(SetConferenceMode, voe_channel());
}
}
send_ = send;
return true;
}
bool WebRtcVoiceMediaChannel::AddStream(uint32 ssrc) {
talk_base::CritScope lock(&mux_channels_cs_);
if (mux_channels_.find(ssrc) != mux_channels_.end()) {
return false;
}
// Create a new channel for receiving audio data.
int channel = engine()->voe()->base()->CreateChannel();
if (channel == -1) {
LOG_RTCERR0(CreateChannel);
return false;
}
// Configure to use external transport, like our default channel.
if (engine()->voe()->network()->RegisterExternalTransport(
channel, *this) == -1) {
LOG_RTCERR2(SetExternalTransport, channel, this);
return false;
}
// Use the same SSRC as our default channel (so the RTCP reports are correct).
unsigned int send_ssrc;
webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) {
LOG_RTCERR2(GetSendSSRC, channel, send_ssrc);
return false;
}
if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) {
LOG_RTCERR2(SetSendSSRC, channel, send_ssrc);
return false;
}
// Use the same recv payload types as our default channel.
ResetRecvCodecs(channel);
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.rate = 0; // Needed to make GetRecPayloadType work for ISAC
if (engine()->voe()->codec()->GetRecPayloadType(
voe_channel(), voe_codec) != -1) {
if (engine()->voe()->codec()->SetRecPayloadType(
channel, voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
return false;
}
}
}
}
if (mux_channels_.empty() && playout_) {
// This is the first stream in a multi user meeting. We can now
// disable playback of the default stream. This since the default
// stream will probably have received some initial packets before
// the new stream was added. This will mean that the CN state from
// the default channel will be mixed in with the other streams
// throughout the whole meeting, which might be disturbing.
LOG(LS_INFO) << "Disabling playback on the default voice channel";
SetPlayout(voe_channel(), false);
}
mux_channels_[ssrc] = channel;
// TODO: We should rollback the add if SetPlayout fails.
LOG(LS_INFO) << "New audio stream " << ssrc
<< " registered to VoiceEngine channel #"
<< channel << ".";
return SetPlayout(channel, playout_);
}
bool WebRtcVoiceMediaChannel::RemoveStream(uint32 ssrc) {
talk_base::CritScope lock(&mux_channels_cs_);
ChannelMap::iterator it = mux_channels_.find(ssrc);
if (it != mux_channels_.end()) {
if (engine()->voe()->network()->DeRegisterExternalTransport(
it->second) == -1) {
LOG_RTCERR1(DeRegisterExternalTransport, it->second);
}
LOG(LS_INFO) << "Removing audio stream " << ssrc
<< " with VoiceEngine channel #"
<< it->second << ".";
if (engine()->voe()->base()->DeleteChannel(it->second) == -1) {
LOG_RTCERR1(DeleteChannel, voe_channel());
return false;
}
mux_channels_.erase(it);
if (mux_channels_.empty() && playout_) {
// The last stream was removed. We can now enable the default
// channel for new channels to be played out immediately without
// waiting for AddStream messages.
// TODO: Does the default channel still have it's CN state?
LOG(LS_INFO) << "Enabling playback on the default voice channel";
SetPlayout(voe_channel(), true);
}
}
return true;
}
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
actives->clear();
for (ChannelMap::iterator it = mux_channels_.begin();
it != mux_channels_.end(); ++it) {
int level = GetOutputLevel(it->second);
if (level > 0) {
actives->push_back(std::make_pair(it->first, level));
}
}
return true;
}
int WebRtcVoiceMediaChannel::GetOutputLevel() {
// return the highest output level of all streams
int highest = GetOutputLevel(voe_channel());
for (ChannelMap::iterator it = mux_channels_.begin();
it != mux_channels_.end(); ++it) {
int level = GetOutputLevel(it->second);
highest = talk_base::_max(level, highest);
}
return highest;
}
bool WebRtcVoiceMediaChannel::SetOutputScaling(
uint32 ssrc, double left, double right) {
talk_base::CritScope lock(&mux_channels_cs_);
// Collect the channels to scale the output volume.
std::vector<int> channels;
if (0 == ssrc) { // Collect all channels, including the default one.
channels.push_back(voe_channel());
for (ChannelMap::const_iterator it = mux_channels_.begin();
it != mux_channels_.end(); ++it) {
channels.push_back(it->second);
}
} else { // Collect only the channel of the specified ssrc.
int channel = GetChannelNum(ssrc);
if (-1 == channel) {
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
return false;
}
channels.push_back(channel);
}
// Scale the output volume for the collected channels. We first normalize to
// scale the volume and then set the left and right pan.
float scale = static_cast<float>(talk_base::_max(left, right));
if (scale > 0.0001f) {
left /= scale;
right /= scale;
}
for (std::vector<int>::const_iterator it = channels.begin();
it != channels.end(); ++it) {
if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(
*it, scale)) {
LOG_RTCERR2(SetChannelOutputVolumeScaling, *it, scale);
return false;
}
if (-1 == engine()->voe()->volume()->SetOutputVolumePan(
*it, static_cast<float>(left), static_cast<float>(right))) {
LOG_RTCERR3(SetOutputVolumePan, *it, left, right);
// Do not return if fails. SetOutputVolumePan is not available for all
// pltforms.
}
LOG(LS_INFO) << "SetOutputScaling to left=" << left * scale
<< " right=" << right * scale
<< " for channel " << *it << " and ssrc " << ssrc;
}
return true;
}
bool WebRtcVoiceMediaChannel::GetOutputScaling(
uint32 ssrc, double* left, double* right) {
if (!left || !right) return false;
talk_base::CritScope lock(&mux_channels_cs_);
// Determine which channel based on ssrc.
int channel = (0 == ssrc) ? voe_channel() : GetChannelNum(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
return false;
}
float scaling;
if (-1 == engine()->voe()->volume()->GetChannelOutputVolumeScaling(
channel, scaling)) {
LOG_RTCERR2(GetChannelOutputVolumeScaling, channel, scaling);
return false;
}
float left_pan;
float right_pan;
if (-1 == engine()->voe()->volume()->GetOutputVolumePan(
channel, left_pan, right_pan)) {
LOG_RTCERR3(GetOutputVolumePan, channel, left_pan, right_pan);
// If GetOutputVolumePan fails, we use the default left and right pan.
left_pan = 1.0f;
right_pan = 1.0f;
}
*left = scaling * left_pan;
*right = scaling * right_pan;
return true;
}
bool WebRtcVoiceMediaChannel::SetRingbackTone(const char *buf, int len) {
ringback_tone_.reset(new WebRtcSoundclipStream(buf, len));
return true;
}
bool WebRtcVoiceMediaChannel::PlayRingbackTone(uint32 ssrc,
bool play, bool loop) {
if (!ringback_tone_.get()) {
return false;
}
// Determine which VoiceEngine channel to play on.
int channel = (ssrc == 0) ? voe_channel() : GetChannelNum(ssrc);
if (channel == -1) {
return false;
}
// Make sure the ringtone is cued properly, and play it out.
if (play) {
ringback_tone_->set_loop(loop);
ringback_tone_->Rewind();
if (engine()->voe()->file()->StartPlayingFileLocally(channel,
ringback_tone_.get()) == -1) {
LOG_RTCERR2(StartPlayingFileLocally, channel, ringback_tone_.get());
LOG(LS_ERROR) << "Unable to start ringback tone";
return false;
}
ringback_channels_.insert(channel);
LOG(LS_INFO) << "Started ringback on channel " << channel;
} else {
if (engine()->voe()->file()->IsPlayingFileLocally(channel) == 1 &&
engine()->voe()->file()->StopPlayingFileLocally(channel) == -1) {
LOG_RTCERR1(StopPlayingFileLocally, channel);
return false;
}
LOG(LS_INFO) << "Stopped ringback on channel " << channel;
ringback_channels_.erase(channel);
}
return true;
}
bool WebRtcVoiceMediaChannel::PressDTMF(int event, bool playout) {
if (!dtmf_allowed_) {
return false;
}
// Enable or disable DTMF playout of this tone as requested. This will linger
// until the next call to this method, but that's OK.
if (engine()->voe()->dtmf()->SetDtmfFeedbackStatus(playout) == -1) {
LOG_RTCERR2(SendDTMF, voe_channel(), playout);
return false;
}
// Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
if (engine()->voe()->dtmf()->SendTelephoneEvent(voe_channel(), event,
true) == -1) {
LOG_RTCERR3(SendDTMF, voe_channel(), event, true);
return false;
}
return true;
}
void WebRtcVoiceMediaChannel::OnPacketReceived(talk_base::Buffer* packet) {
// Pick which channel to send this packet to. If this packet doesn't match
// any multiplexed streams, just send it to the default channel. Otherwise,
// send it to the specific decoder instance for that stream.
int which_channel = GetChannelNum(
ParseSsrc(packet->data(), packet->length(), false));
if (which_channel == -1) {
which_channel = voe_channel();
}
// Stop any ringback that might be playing on the channel.
// It's possible the ringback has already stopped, ih which case we'll just
// use the opportunity to remove the channel from ringback_channels_.
const std::set<int>::iterator it = ringback_channels_.find(which_channel);
if (it != ringback_channels_.end()) {
if (engine()->voe()->file()->IsPlayingFileLocally(
which_channel) == 1) {
engine()->voe()->file()->StopPlayingFileLocally(which_channel);
LOG(LS_INFO) << "Stopped ringback on channel " << which_channel
<< " due to incoming media";
}
ringback_channels_.erase(which_channel);
}
// Pass it off to the decoder.
engine()->voe()->network()->ReceivedRTPPacket(which_channel,
packet->data(),
packet->length());
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(talk_base::Buffer* packet) {
// See above.
int which_channel = GetChannelNum(
ParseSsrc(packet->data(), packet->length(), true));
if (which_channel == -1) {
which_channel = voe_channel();
}
engine()->voe()->network()->ReceivedRTCPPacket(which_channel,
packet->data(),
packet->length());
}
void WebRtcVoiceMediaChannel::SetSendSsrc(uint32 ssrc) {
if (engine()->voe()->rtp()->SetLocalSSRC(voe_channel(), ssrc)
== -1) {
LOG_RTCERR2(SetSendSSRC, voe_channel(), ssrc);
}
}
bool WebRtcVoiceMediaChannel::SetRtcpCName(const std::string& cname) {
if (engine()->voe()->rtp()->SetRTCP_CNAME(voe_channel(),
cname.c_str()) == -1) {
LOG_RTCERR2(SetRTCP_CNAME, voe_channel(), cname);
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::Mute(bool muted) {
if (engine()->voe()->volume()->SetInputMute(voe_channel(),
muted) == -1) {
LOG_RTCERR2(SetInputMute, voe_channel(), muted);
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
// In VoiceEngine 3.5, GetRTCPStatistics will return 0 even when it fails,
// causing the stats to contain garbage information. To prevent this, we
// zero the stats structure before calling this API.
// TODO: Remove this workaround.
webrtc::CallStatistics cs;
unsigned int ssrc;
webrtc::CodecInst codec;
unsigned int level;
// Fill in the sender info, based on what we know, and what the
// remote side told us it got from its RTCP report.
VoiceSenderInfo sinfo;
// Data we obtain locally.
memset(&cs, 0, sizeof(cs));
if (engine()->voe()->rtp()->GetRTCPStatistics(voe_channel(), cs) == -1 ||
engine()->voe()->rtp()->GetLocalSSRC(voe_channel(), ssrc) == -1) {
return false;
}
sinfo.ssrc = ssrc;
sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
sinfo.bytes_sent = cs.bytesSent;
sinfo.packets_sent = cs.packetsSent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
// Data from the last remote RTCP report.
unsigned int ntp_high, ntp_low, timestamp, ptimestamp, jitter;
unsigned short loss; // NOLINT
if (engine()->voe()->rtp()->GetRemoteRTCPData(voe_channel(),
ntp_high, ntp_low, timestamp, ptimestamp, &jitter, &loss) != -1 &&
engine()->voe()->codec()->GetSendCodec(voe_channel(),
codec) != -1) {
// Convert Q8 to floating point.
sinfo.fraction_lost = static_cast<float>(loss) / (1 << 8);
// Convert samples to milliseconds.
if (codec.plfreq / 1000 > 0) {
sinfo.jitter_ms = jitter / (codec.plfreq / 1000);
}
} else {
sinfo.fraction_lost = -1;
sinfo.jitter_ms = -1;
}
// TODO: Figure out how to get remote packets_lost, ext_seqnum
sinfo.packets_lost = -1;
sinfo.ext_seqnum = -1;
// Local speech level.
sinfo.audio_level = (engine()->voe()->volume()->
GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
bool echo_metrics_on = false;
// These can take on valid negative values, so use the lowest possible level
// as default rather than -1.
sinfo.echo_return_loss = -100;
sinfo.echo_return_loss_enhancement = -100;
// These can also be negative, but in practice -1 is only used to signal
// insufficient data, since the resolution is limited to multiples of 4 ms.
sinfo.echo_delay_median_ms = -1;
sinfo.echo_delay_std_ms = -1;
if (engine()->voe()->processing()->GetEcMetricsStatus(echo_metrics_on) !=
-1 && echo_metrics_on) {
// TODO: we may want to use VoECallReport::GetEchoMetricsSummary
// here, but it appears to be unsuitable currently. Revisit after this is
// investigated: http://b/issue?id=5666755
int erl, erle, rerl, anlp;
if (engine()->voe()->processing()->GetEchoMetrics(erl, erle, rerl, anlp) !=
-1) {
sinfo.echo_return_loss = erl;
sinfo.echo_return_loss_enhancement = erle;
}
int median, std;
if (engine()->voe()->processing()->GetEcDelayMetrics(median, std) != -1) {
sinfo.echo_delay_median_ms = median;
sinfo.echo_delay_std_ms = std;
}
}
info->senders.push_back(sinfo);
// Build the list of receivers, one for each mux channel, or 1 in a 1:1 call.
std::vector<int> channels;
for (ChannelMap::const_iterator it = mux_channels_.begin();
it != mux_channels_.end(); ++it) {
channels.push_back(it->second);
}
if (channels.empty()) {
channels.push_back(voe_channel());
}
// Get the SSRC and stats for each receiver, based on our own calculations.
for (std::vector<int>::const_iterator it = channels.begin();
it != channels.end(); ++it) {
memset(&cs, 0, sizeof(cs));
if (engine()->voe()->rtp()->GetRemoteSSRC(*it, ssrc) != -1 &&
engine()->voe()->rtp()->GetRTCPStatistics(*it, cs) != -1 &&
engine()->voe()->codec()->GetRecCodec(*it, codec) != -1) {
VoiceReceiverInfo rinfo;
rinfo.ssrc = ssrc;
rinfo.bytes_rcvd = cs.bytesReceived;
rinfo.packets_rcvd = cs.packetsReceived;
// The next four fields are from the most recently sent RTCP report.
// Convert Q8 to floating point.
rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
rinfo.packets_lost = cs.cumulativeLost;
rinfo.ext_seqnum = cs.extendedMax;
// Convert samples to milliseconds.
if (codec.plfreq / 1000 > 0) {
rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
}
// Get jitter buffer and total delay (alg + jitter + playout) stats.
webrtc::NetworkStatistics ns;
if (engine()->voe()->neteq() &&
engine()->voe()->neteq()->GetNetworkStatistics(
*it, ns) != -1) {
rinfo.jitter_buffer_ms = ns.currentBufferSize;
rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
}
if (engine()->voe()->sync()) {
engine()->voe()->sync()->GetDelayEstimate(*it,
rinfo.delay_estimate_ms);
}
// Get speech level.
rinfo.audio_level = (engine()->voe()->volume()->
GetSpeechOutputLevelFullRange(*it, level) != -1) ? level : -1;
info->receivers.push_back(rinfo);
}
}
return true;
}
void WebRtcVoiceMediaChannel::GetLastMediaError(
uint32* ssrc, VoiceMediaChannel::Error* error) {
ASSERT(ssrc != NULL);
ASSERT(error != NULL);
FindSsrc(voe_channel(), ssrc);
*error = WebRtcErrorToChannelError(GetLastEngineError());
}
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
talk_base::CritScope lock(&mux_channels_cs_);
ASSERT(ssrc != NULL);
if (channel_num == voe_channel()) {
unsigned local_ssrc = 0;
// This is a sending channel.
if (engine()->voe()->rtp()->GetLocalSSRC(
channel_num, local_ssrc) != -1) {
*ssrc = local_ssrc;
}
return true;
} else if (channel_num == -1 && send_ != SEND_NOTHING) {
// Sometimes the VoiceEngine core will throw error with channel_num = -1.
// This means the error is not limited to a specific channel. Signal the
// message using ssrc=0. If the current channel is sending, use this
// channel for sending the message.
*ssrc = 0;
return true;
} else {
// Check whether this is a receiving channel.
for (ChannelMap::const_iterator it = mux_channels_.begin();
it != mux_channels_.end(); ++it) {
if (it->second == channel_num) {
*ssrc = it->first;
return true;
}
}
}
return false;
}
void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
SignalMediaError(ssrc, WebRtcErrorToChannelError(error));
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
unsigned int ulevel;
int ret =
engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
return (ret == 0) ? static_cast<int>(ulevel) : -1;
}
int WebRtcVoiceMediaChannel::GetChannelNum(uint32 ssrc) {
ChannelMap::iterator it = mux_channels_.find(ssrc);
return (it != mux_channels_.end()) ? it->second : -1;
}
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
// Get the RED encodings from the parameter with no name. This may
// change based on what is discussed on the Jingle list.
// The encoding parameter is of the form "a/b"; we only support where
// a == b. Verify this and parse out the value into red_pt.
// If the parameter value is absent (as it will be until we wire up the
// signaling of this message), use the second codec specified (i.e. the
// one after "red") as the encoding parameter.
int red_pt = -1;
std::string red_params;
CodecParameterMap::const_iterator it = red_codec.params.find("");
if (it != red_codec.params.end()) {
red_params = it->second;
std::vector<std::string> red_pts;
if (talk_base::split(red_params, '/', &red_pts) != 2 ||
red_pts[0] != red_pts[1] ||
!talk_base::FromString(red_pts[0], &red_pt)) {
LOG(LS_WARNING) << "RED params " << red_params << " not supported.";
return false;
}
} else if (red_codec.params.empty()) {
LOG(LS_WARNING) << "RED params not present, using defaults";
if (all_codecs.size() > 1) {
red_pt = all_codecs[1].id;
}
}
// Try to find red_pt in |codecs|.
std::vector<AudioCodec>::const_iterator codec;
for (codec = all_codecs.begin(); codec != all_codecs.end(); ++codec) {
if (codec->id == red_pt)
break;
}
// If we find the right codec, that will be the codec we pass to
// SetSendCodec, with the desired payload type.
if (codec != all_codecs.end() &&
engine()->FindWebRtcCodec(*codec, send_codec)) {
} else {
LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::EnableRtcp(int channel) {
if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
LOG_RTCERR2(SetRTCPStatus, voe_channel(), 1);
return false;
}
// TODO: Enable VQMon and RTCP XR reports, once we know what
// what we want to do with them.
// engine()->voe().EnableVQMon(voe_channel(), true);
// engine()->voe().EnableRTCP_XR(voe_channel(), true);
return true;
}
bool WebRtcVoiceMediaChannel::ResetRecvCodecs(int channel) {
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;
}
}
}
return true;
}
bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
if (playout) {
LOG(LS_INFO) << "Starting playout for channel #" << channel;
if (engine()->voe()->base()->StartPlayout(channel) == -1) {
LOG_RTCERR1(StartPlayout, channel);
return false;
}
} else {
LOG(LS_INFO) << "Stopping playout for channel #" << channel;
engine()->voe()->base()->StopPlayout(channel);
}
return true;
}
uint32 WebRtcVoiceMediaChannel::ParseSsrc(const void* data, size_t len,
bool rtcp) {
size_t ssrc_pos = (!rtcp) ? 8 : 4;
uint32 ssrc = 0;
if (len >= (ssrc_pos + sizeof(ssrc))) {
ssrc = talk_base::GetBE32(static_cast<const char*>(data) + ssrc_pos);
}
return ssrc;
}
// 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;
}
}
int WebRtcSoundclipStream::Read(void *buf, int len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
return res;
}
int WebRtcSoundclipStream::Rewind() {
mem_.Rewind();
// Return -1 to keep VoiceEngine from looping.
return (loop_) ? 0 : -1;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE