blob: 0e964729e431086c41b07e3b9c429acd3ca2d7f6 [file] [log] [blame]
/*
* libjingle
* Copyright 2004 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/media/webrtc/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/media/base/audiorenderer.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/voiceprocessor.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#ifdef WIN32
#include <objbase.h> // NOLINT
#endif
namespace cricket {
struct CodecPref {
const char* name;
int clockrate;
int channels;
int payload_type;
bool is_multi_rate;
};
static const CodecPref kCodecPrefs[] = {
{ "OPUS", 48000, 2, 111, true },
{ "ISAC", 16000, 1, 103, true },
{ "ISAC", 32000, 1, 104, true },
{ "CELT", 32000, 1, 109, true },
{ "CELT", 32000, 2, 110, true },
{ "G722", 16000, 1, 9, false },
{ "ILBC", 8000, 1, 102, false },
{ "PCMU", 8000, 1, 0, false },
{ "PCMA", 8000, 1, 8, false },
{ "CN", 48000, 1, 107, false },
{ "CN", 32000, 1, 106, false },
{ "CN", 16000, 1, 105, false },
{ "CN", 8000, 1, 13, false },
{ "red", 8000, 1, 127, false },
{ "telephone-event", 8000, 1, 126, false },
};
// 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 int kRtpAudioLevelHeaderExtensionId = 1;
static const char kIsacCodecName[] = "ISAC";
static const char kL16CodecName[] = "L16";
// Codec parameters for Opus.
static const int kOpusMonoBitrate = 32000;
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
static const int kNackMaxPackets = 250;
static const int kOpusStereoBitrate = 64000;
// draft-spittka-payload-rtp-opus-03
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
// Ensure we open the file in a writeable path on ChromeOS and Android. This
// workaround can be removed when it's possible to specify a filename for audio
// option based AEC dumps.
//
// TODO(grunell): Use a string in the options instead of hardcoding it here
// and let the embedder choose the filename (crbug.com/264223).
//
// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
// below.
#if defined(CHROMEOS)
static const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
#elif defined(ANDROID)
static const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
#else
static const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
#endif
// 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;
}
}
// Severity is an integer because it comes is assumed to be from command line.
static int SeverityToFilter(int severity) {
int filter = webrtc::kTraceNone;
switch (severity) {
case talk_base::LS_VERBOSE:
filter |= webrtc::kTraceAll;
case talk_base::LS_INFO:
filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
case talk_base::LS_WARNING:
filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
case talk_base::LS_ERROR:
filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
}
return filter;
}
static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
if (_stricmp(kCodecPrefs[i].name, codec.plname) == 0 &&
kCodecPrefs[i].clockrate == codec.plfreq) {
return kCodecPrefs[i].is_multi_rate;
}
}
return false;
}
static bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec) {
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
if (it->Matches(codec)) {
if (found_codec != NULL) {
*found_codec = *it;
}
return true;
}
}
return false;
}
static bool IsNackEnabled(const AudioCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
}
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) {
// The voe file api is not available in chrome.
if (!engine_->voe_sc()->file()) {
return false;
}
// 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_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
tx_processor_ssrc_(0),
rx_processor_ssrc_(0) {
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_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
tx_processor_ssrc_(0),
rx_processor_ssrc_(0) {
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,
kRtpAudioLevelHeaderExtensionId));
}
static bool IsOpus(const AudioCodec& codec) {
return (_stricmp(codec.name.c_str(), kOpusCodecName) == 0);
}
static bool IsIsac(const AudioCodec& codec) {
return (_stricmp(codec.name.c_str(), kIsacCodecName) == 0);
}
// True if params["stereo"] == "1"
static bool IsOpusStereoEnabled(const AudioCodec& codec) {
CodecParameterMap::const_iterator param =
codec.params.find(kCodecParamStereo);
if (param == codec.params.end()) {
return false;
}
return param->second == kParamValueTrue;
}
static bool IsValidOpusBitrate(int bitrate) {
return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate);
}
// Returns 0 if params[kCodecParamMaxAverageBitrate] is not defined or invalid.
// Returns the value of params[kCodecParamMaxAverageBitrate] otherwise.
static int GetOpusBitrateFromParams(const AudioCodec& codec) {
int bitrate = 0;
if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
return 0;
}
if (!IsValidOpusBitrate(bitrate)) {
LOG(LS_WARNING) << "Codec parameter \"maxaveragebitrate\" has an "
<< "invalid value: " << bitrate;
return 0;
}
return bitrate;
}
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 &&
kCodecPrefs[j].channels == voe_codec.channels) {
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);
if (IsIsac(codec)) {
// Indicate auto-bandwidth in signaling.
codec.bitrate = 0;
}
if (IsOpus(codec)) {
// Only add fmtp parameters that differ from the spec.
if (kPreferredMinPTime != kOpusDefaultMinPTime) {
codec.params[kCodecParamMinPTime] =
talk_base::ToString(kPreferredMinPTime);
}
if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
codec.params[kCodecParamMaxPTime] =
talk_base::ToString(kPreferredMaxPTime);
}
// TODO(hellner): Add ptime, sprop-stereo, stereo and useinbandfec
// when they can be set to values other than the default.
}
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(talk_base::Thread* worker_thread) {
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;
}
// Gets the default set of optoins 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.
static 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.typing_detection.Set(true);
options.conference_mode.Set(false);
options.adjust_agc_delta.Set(0);
options.experimental_agc.Set(false);
options.experimental_aec.Set(false);
options.aec_dump.Set(false);
return options;
}
bool WebRtcVoiceEngine::InitInternal() {
// Temporarily turn logging level up for the Init call
int old_filter = log_filter_;
int extended_filter = log_filter_ | SeverityToFilter(talk_base::LS_INFO);
SetTraceFilter(extended_filter);
SetTraceOptions("");
// Init WebRtc VoiceEngine.
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(talk_base::LS_INFO, buffer);
// Save the default AGC configuration settings. This must happen before
// calling SetOptions 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 (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(juberti): Clean this up when we support setting the soundclip device.
#ifdef WIN32
// The SetPlayoutDevice may not be implemented in the case of external ADM.
// TODO(ronghuawu): We should only check the adm_sc_ here, but current
// PeerConnection interface never set the adm_sc_, so need to check both
// in order to determine if the external adm is used.
if (!adm_ && !adm_sc_) {
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
// 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);
}
initialized_ = true;
return true;
}
void WebRtcVoiceEngine::Terminate() {
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
initialized_ = false;
StopAecDump();
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(const AudioOptions& options) {
if (!ApplyOptions(options)) {
return false;
}
options_ = options;
return true;
}
bool WebRtcVoiceEngine::SetOptionOverrides(const AudioOptions& overrides) {
LOG(LS_INFO) << "Setting option overrides: " << overrides.ToString();
if (!ApplyOptions(overrides)) {
return false;
}
option_overrides_ = overrides;
return true;
}
bool WebRtcVoiceEngine::ClearOptionOverrides() {
LOG(LS_INFO) << "Clearing option overrides.";
AudioOptions options = options_;
// Only call ApplyOptions if |options_overrides_| contains overrided options.
// ApplyOptions affects NS, AGC other options that is shared between
// all WebRtcVoiceEngineChannels.
if (option_overrides_ == AudioOptions()) {
return true;
}
if (!ApplyOptions(options)) {
return false;
}
option_overrides_ = AudioOptions();
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) {
AudioOptions options = options_in; // The options are modified below.
// 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 defined(IOS)
// On iOS, VPIO provides built-in EC and AGC.
options.echo_cancellation.Set(false);
options.auto_gain_control.Set(false);
#elif defined(ANDROID)
ec_mode = webrtc::kEcAecm;
#endif
#if defined(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.experimental_aec.Set(false);
#endif
LOG(LS_INFO) << "Applying audio options: " << options.ToString();
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
bool echo_cancellation;
if (options.echo_cancellation.Get(&echo_cancellation)) {
if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) {
LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode);
return false;
}
#if !defined(ANDROID)
// TODO(ajm): Remove the error return on Android from webrtc.
if (voep->SetEcMetricsStatus(echo_cancellation) == -1) {
LOG_RTCERR1(SetEcMetricsStatus, 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);
return false;
}
}
}
bool auto_gain_control;
if (options.auto_gain_control.Get(&auto_gain_control)) {
if (voep->SetAgcStatus(auto_gain_control, agc_mode) == -1) {
LOG_RTCERR2(SetAgcStatus, auto_gain_control, agc_mode);
return false;
}
}
bool noise_suppression;
if (options.noise_suppression.Get(&noise_suppression)) {
if (voep->SetNsStatus(noise_suppression, ns_mode) == -1) {
LOG_RTCERR2(SetNsStatus, noise_suppression, ns_mode);
return false;
}
}
bool highpass_filter;
if (options.highpass_filter.Get(&highpass_filter)) {
if (voep->EnableHighPassFilter(highpass_filter) == -1) {
LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter);
return false;
}
}
bool stereo_swapping;
if (options.stereo_swapping.Get(&stereo_swapping)) {
voep->EnableStereoChannelSwapping(stereo_swapping);
if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) {
LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping);
return false;
}
}
bool typing_detection;
if (options.typing_detection.Get(&typing_detection)) {
if (voep->SetTypingDetectionStatus(typing_detection) == -1) {
// In case of error, log the info and continue
LOG_RTCERR1(SetTypingDetectionStatus, typing_detection);
}
}
int adjust_agc_delta;
if (options.adjust_agc_delta.Get(&adjust_agc_delta)) {
if (!AdjustAgcLevel(adjust_agc_delta)) {
return false;
}
}
bool aec_dump;
if (options.aec_dump.Get(&aec_dump)) {
if (aec_dump)
StartAecDump(kAecDumpByAudioOptionFilename);
else
StopAecDump();
}
bool experimental_aec;
if (options.experimental_aec.Get(&experimental_aec)) {
webrtc::AudioProcessing* audioproc =
voe_wrapper_->base()->audio_processing();
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
if (audioproc) {
webrtc::Config config;
config.Set<webrtc::DelayCorrection>(
new webrtc::DelayCorrection(experimental_aec));
audioproc->SetExtraOptions(config);
}
}
return true;
}
bool WebRtcVoiceEngine::SetDelayOffset(int offset) {
voe_wrapper_->processing()->SetDelayOffsetMs(offset);
if (voe_wrapper_->processing()->DelayOffsetMs() != offset) {
LOG_RTCERR1(SetDelayOffsetMs, offset);
return false;
}
return true;
}
struct ResumeEntry {
ResumeEntry(WebRtcVoiceMediaChannel *c, bool p, SendFlags s)
: channel(c),
playout(p),
send(s) {
}
WebRtcVoiceMediaChannel *channel;
bool playout;
SendFlags send;
};
// 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) && !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) {
// The voe file api is not available in chrome.
if (!voe_wrapper_->file()) {
return false;
}
if (enable && !monitor_) {
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_) {
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);
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;
}
// Apply codec-specific settings.
if (IsIsac(codec)) {
// If ISAC and an explicit bitrate is not specified,
// enable auto bandwidth 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_ : "");
}
int WebRtcVoiceEngine::GetLastEngineError() {
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;
talk_base::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);
}
}
// 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();
}
}
// 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 statistics available",
"TransmitMixer::TypingDetection() VE_TYPING_NOISE_WARNING message has been posted", // NOLINT
"TransmitMixer::TypingDetection() pending noise-saturation warning exists", // NOLINT
"GetRecPayloadType() failed to retrieve RX payload type (error=10026)", // NOLINT
"StopPlayingFileAsMicrophone() isnot playing (error=8088)",
NULL
};
for (const char* const* p = kTracesToIgnore; *p; ++p) {
if (trace.find(*p) != std::string::npos) {
return true;
}
}
return false;
}
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
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;
else if (level == webrtc::kTraceTerseInfo)
sev = talk_base::LS_INFO;
// 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(int channel_num, 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);
ASSERT(direction == MPD_RX || direction == MPD_TX);
*channel_num = -1;
// Find corresponding channel for ssrc.
for (ChannelList::const_iterator it = channels_.begin();
it != channels_.end(); ++it) {
ASSERT(*it != NULL);
if (direction & MPD_RX) {
*channel_num = (*it)->GetReceiveChannelNum(ssrc);
}
if (*channel_num == -1 && (direction & MPD_TX)) {
*channel_num = (*it)->GetSendChannelNum(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;
}
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;
uint32* processor_ssrc = NULL;
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;
}
webrtc::ProcessingTypes processing_type;
{
talk_base::CritScope cs(&signal_media_critical_);
if (direction == MPD_RX) {
processing_type = webrtc::kPlaybackAllChannelsMixed;
if (SignalRxMediaFrame.is_empty()) {
register_with_webrtc = true;
processor_ssrc = &rx_processor_ssrc_;
}
SignalRxMediaFrame.connect(voice_processor,
&VoiceProcessor::OnFrame);
} else {
processing_type = webrtc::kRecordingPerChannel;
if (SignalTxMediaFrame.is_empty()) {
register_with_webrtc = true;
processor_ssrc = &tx_processor_ssrc_;
}
SignalTxMediaFrame.connect(voice_processor,
&VoiceProcessor::OnFrame);
}
}
if (register_with_webrtc) {
// TODO(janahan): when registering consider instantiating a
// a VoeMediaProcess object and not make the engine extend the interface.
if (voe()->media() && voe()->media()->
RegisterExternalMediaProcessing(channel_id,
processing_type,
*this) != -1) {
LOG(LS_INFO) << "Media Processing Registration Succeeded. channel:"
<< channel_id;
*processor_ssrc = ssrc;
success = true;
} else {
LOG_RTCERR2(RegisterExternalMediaProcessing,
channel_id,
processing_type);
success = false;
}
} 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::UnregisterProcessorChannel(
MediaProcessorDirection channel_direction,
uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection processor_direction) {
bool success = true;
FrameSignal* signal;
webrtc::ProcessingTypes processing_type;
uint32* processor_ssrc = NULL;
if (channel_direction == MPD_RX) {
signal = &SignalRxMediaFrame;
processing_type = webrtc::kPlaybackAllChannelsMixed;
processor_ssrc = &rx_processor_ssrc_;
} else {
signal = &SignalTxMediaFrame;
processing_type = webrtc::kRecordingPerChannel;
processor_ssrc = &tx_processor_ssrc_;
}
int deregister_id = -1;
{
talk_base::CritScope cs(&signal_media_critical_);
if ((processor_direction & channel_direction) != 0 && !signal->is_empty()) {
signal->disconnect(voice_processor);
int channel_id = -1;
bool found_channel = FindChannelNumFromSsrc(ssrc,
channel_direction,
&channel_id);
if (signal->is_empty() && found_channel) {
deregister_id = channel_id;
}
}
}
if (deregister_id != -1) {
if (voe()->media() &&
voe()->media()->DeRegisterExternalMediaProcessing(deregister_id,
processing_type) != -1) {
*processor_ssrc = 0;
LOG(LS_INFO) << "Media Processing DeRegistration Succeeded. channel:"
<< deregister_id;
} else {
LOG_RTCERR2(DeRegisterExternalMediaProcessing,
deregister_id,
processing_type);
success = false;
}
}
return success;
}
bool WebRtcVoiceEngine::UnregisterProcessor(
uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection direction) {
bool success = true;
if (voice_processor == NULL) {
LOG(LS_WARNING) << "Media Processing Deregistration Failed. ssrc: "
<< ssrc;
return false;
}
if (!UnregisterProcessorChannel(MPD_RX, ssrc, voice_processor, direction)) {
success = false;
}
if (!UnregisterProcessorChannel(MPD_TX, ssrc, voice_processor, direction)) {
success = false;
}
return success;
}
// Implementing method from WebRtc VoEMediaProcess interface
// Do not lock mux_channel_cs_ in this callback.
void WebRtcVoiceEngine::Process(int channel,
webrtc::ProcessingTypes type,
int16_t audio10ms[],
int length,
int sampling_freq,
bool is_stereo) {
talk_base::CritScope cs(&signal_media_critical_);
AudioFrame frame(audio10ms, length, sampling_freq, is_stereo);
if (type == webrtc::kPlaybackAllChannelsMixed) {
SignalRxMediaFrame(rx_processor_ssrc_, MPD_RX, &frame);
} else if (type == webrtc::kRecordingPerChannel) {
SignalTxMediaFrame(tx_processor_ssrc_, MPD_TX, &frame);
} else {
LOG(LS_WARNING) << "Media Processing invoked unexpectedly."
<< " channel: " << channel << " type: " << type
<< " tx_ssrc: " << tx_processor_ssrc_
<< " rx_ssrc: " << rx_processor_ssrc_;
}
}
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
if (!is_dumping_aec_) {
// Start dumping AEC when we are not dumping.
if (voe_wrapper_->processing()->StartDebugRecording(
filename.c_str()) != webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StartDebugRecording);
} else {
is_dumping_aec_ = true;
}
}
}
void WebRtcVoiceEngine::StopAecDump() {
if (is_dumping_aec_) {
// Stop dumping AEC when we are dumping.
if (voe_wrapper_->processing()->StopDebugRecording() !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StopDebugRecording);
}
is_dumping_aec_ = false;
}
}
// This struct relies on the generated copy constructor and assignment operator
// since it is used in an stl::map.
struct WebRtcVoiceMediaChannel::WebRtcVoiceChannelInfo {
WebRtcVoiceChannelInfo() : channel(-1), renderer(NULL) {}
WebRtcVoiceChannelInfo(int ch, AudioRenderer* r)
: channel(ch),
renderer(r) {}
~WebRtcVoiceChannelInfo() {}
int channel;
AudioRenderer* renderer;
};
// WebRtcVoiceMediaChannel
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->voe()->base()->CreateChannel()),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
nack_enabled_(false),
playout_(false),
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
<< voe_channel();
ConfigureSendChannel(voe_channel());
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
// Remove any remaining send streams, the default channel will be deleted
// later.
while (!send_channels_.empty())
RemoveSendStream(send_channels_.begin()->first);
// Unregister ourselves from the engine.
engine()->UnregisterChannel(this);
// Remove any remaining streams.
while (!receive_channels_.empty()) {
RemoveRecvStream(receive_channels_.begin()->first);
}
// Delete the default channel.
DeleteChannel(voe_channel());
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
// TODO(xians): Add support to set different options for different send
// streams after we support multiple APMs.
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
options_.SetAll(options);
if (send_ != SEND_NOTHING) {
if (!engine()->SetOptionOverrides(options_)) {
LOG(LS_WARNING) <<
"Failed to engine SetOptionOverrides during channel SetOptions.";
return false;
}
} else {
// Will be interpreted when appropriate.
}
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
}
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
// Set the payload types to be used for incoming media.
LOG(LS_INFO) << "Setting receive voice codecs:";
std::vector<AudioCodec> new_codecs;
// Find all new codecs. We allow adding new codecs but don't allow changing
// the payload type of codecs that is already configured since we might
// already be receiving packets with that payload type.
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
AudioCodec old_codec;
if (FindCodec(recv_codecs_, *it, &old_codec)) {
if (old_codec.id != it->id) {
LOG(LS_ERROR) << it->name << " payload type changed.";
return false;
}
} else {
new_codecs.push_back(*it);
}
}
if (new_codecs.empty()) {
// There are no new codecs to configure. Already configured codecs are
// never removed.
return true;
}
if (playout_) {
// Receive codecs can not be changed while playing. So we temporarily
// pause playout.
PausePlayout();
}
bool ret = true;
for (std::vector<AudioCodec>::const_iterator it = new_codecs.begin();
it != new_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 (default_receive_ssrc_ == 0) {
// Set the receive codecs on the default channel explicitly if the
// default channel is not used by |receive_channels_|, this happens in
// conference mode or in non-conference mode when there is no playout
// channel.
// TODO(xians): Figure out how we use the default channel in conference
// mode.
if (engine()->voe()->codec()->SetRecPayloadType(
voe_channel(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, voe_channel(), ToString(voe_codec));
ret = false;
}
}
// Set the receive codecs on all receiving channels.
for (ChannelMap::iterator it = receive_channels_.begin();
it != receive_channels_.end() && ret; ++it) {
if (engine()->voe()->codec()->SetRecPayloadType(
it->second.channel, voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, it->second.channel,
ToString(voe_codec));
ret = false;
}
}
} else {
LOG(LS_WARNING) << "Unknown codec " << ToString(*it);
ret = false;
}
}
if (ret) {
recv_codecs_ = codecs;
}
if (desired_playout_ && !playout_) {
ResumePlayout();
}
return ret;
}
bool WebRtcVoiceMediaChannel::SetSendCodecs(
int channel, const std::vector<AudioCodec>& codecs) {
// Disable VAD, and FEC unless we know the other side wants them.
engine()->voe()->codec()->SetVADStatus(channel, false);
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
engine()->voe()->rtp()->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.
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;
}
// If OPUS, change what we send according to the "stereo" codec
// parameter, and not the "channels" parameter. We set
// voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
// the bitrate is not specified, i.e. is zero, we set it to the
// appropriate default value for mono or stereo Opus.
if (IsOpus(*it)) {
if (IsOpusStereoEnabled(*it)) {
voe_codec.channels = 2;
if (!IsValidOpusBitrate(it->bitrate)) {
if (it->bitrate != 0) {
LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
<< it->bitrate
<< ") with default opus stereo bitrate: "
<< kOpusStereoBitrate;
}
voe_codec.rate = kOpusStereoBitrate;
}
} else {
voe_codec.channels = 1;
if (!IsValidOpusBitrate(it->bitrate)) {
if (it->bitrate != 0) {
LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
<< it->bitrate
<< ") with default opus mono bitrate: "
<< kOpusMonoBitrate;
}
voe_codec.rate = kOpusMonoBitrate;
}
}
int bitrate_from_params = GetOpusBitrateFromParams(*it);
if (bitrate_from_params != 0) {
voe_codec.rate = bitrate_from_params;
}
}
// Find the DTMF telephone event "codec" and tell VoiceEngine channels
// about it.
if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
_stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
channel, it->id) == -1) {
LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, it->id);
return false;
}
}
// 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;
}
// Set the CN payloadtype and the VAD status.
// The CN payload type for 8000 Hz clockrate is fixed at 13.
if (cn_freq != webrtc::kFreq8000Hz) {
if (engine()->voe()->codec()->SetSendCNPayloadType(
channel, it->id, cn_freq) == -1) {
LOG_RTCERR3(SetSendCNPayloadType, channel, it->id, cn_freq);
// TODO(ajm): This failure condition will be removed from VoE.
// Restore the return here when we update to a new enough webrtc.
//
// Not returning false because the SetSendCNPayloadType will fail if
// the channel is already sending.
// This can happen if the remote description is applied twice, for
// example in the case of ROAP on top of JSEP, where both side will
// send the offer.
}
}
// Only turn on VAD if we have a CN payload type that matches the
// clockrate for the codec we are going to use.
if (it->clockrate == send_codec.plfreq) {
LOG(LS_INFO) << "Enabling VAD";
if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
LOG_RTCERR2(SetVADStatus, channel, true);
return false;
}
}
}
// 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(channel, true, it->id) == -1) {
LOG_RTCERR3(SetFECStatus, channel, true, it->id);
return false;
}
} else {
send_codec = voe_codec;
nack_enabled_ = IsNackEnabled(*it);
SetNack(channel, nack_enabled_);
}
first = false;
// Set the codec immediately, since SetVADStatus() depends on whether
// the current codec is mono or stereo.
if (!SetSendCodec(channel, send_codec))
return 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);
if (!SetSendCodec(channel, send_codec))
return false;
}
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
return true;
}
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
dtmf_allowed_ = false;
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
// Find the DTMF telephone event "codec".
if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
_stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
dtmf_allowed_ = true;
}
}
// Cache the codecs in order to configure the channel created later.
send_codecs_ = codecs;
for (ChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (!SetSendCodecs(iter->second.channel, codecs)) {
return false;
}
}
SetNack(receive_channels_, nack_enabled_);
return true;
}
void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels,
bool nack_enabled) {
for (ChannelMap::const_iterator it = channels.begin();
it != channels.end(); ++it) {
SetNack(it->second.channel, nack_enabled);
}
}
void WebRtcVoiceMediaChannel::SetNack(int channel, bool nack_enabled) {
if (nack_enabled) {
LOG(LS_INFO) << "Enabling NACK for channel " << channel;
engine()->voe()->rtp()->SetNACKStatus(channel, true, kNackMaxPackets);
} else {
LOG(LS_INFO) << "Disabling NACK for channel " << channel;
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
}
}
bool WebRtcVoiceMediaChannel::SetSendCodec(
const webrtc::CodecInst& send_codec) {
LOG(LS_INFO) << "Selected voice codec " << ToString(send_codec)
<< ", bitrate=" << send_codec.rate;
for (ChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (!SetSendCodec(iter->second.channel, send_codec))
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetSendCodec(
int channel, const webrtc::CodecInst& send_codec) {
LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
<< ToString(send_codec) << ", bitrate=" << send_codec.rate;
if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) {
LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
return false;
}
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;
}
}
LOG(LS_INFO) << "Enabling audio level header extension with ID " << id;
for (ChannelMap::const_iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus(
iter->second.channel, enable, id) == -1) {
LOG_RTCERR3(SetRTPAudioLevelIndicationStatus,
iter->second.channel, enable, id);
return false;
}
}
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;
}
// Change the playout of all channels to the new state.
bool result = true;
if (receive_channels_.empty()) {
// Only toggle the default channel if we don't have any other channels.
result = SetPlayout(voe_channel(), playout);
}
for (ChannelMap::iterator it = receive_channels_.begin();
it != receive_channels_.end() && result; ++it) {
if (!SetPlayout(it->second.channel, playout)) {
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
<< it->second.channel << " failed";
result = false;
}
}
if (result) {
playout_ = playout;
}
return result;
}
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
desired_send_ = send;
if (!send_channels_.empty())
return ChangeSend(desired_send_);
return true;
}
bool WebRtcVoiceMediaChannel::PauseSend() {
return ChangeSend(SEND_NOTHING);
}
bool WebRtcVoiceMediaChannel::ResumeSend() {
return ChangeSend(desired_send_);
}
bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
if (send_ == send) {
return true;
}
// Change the settings on each send channel.
if (send == SEND_MICROPHONE)
engine()->SetOptionOverrides(options_);
// Change the settings on each send channel.
for (ChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (!ChangeSend(iter->second.channel, send))
return false;
}
// Clear up the options after stopping sending.
if (send == SEND_NOTHING)
engine()->ClearOptionOverrides();
send_ = send;
return true;
}
bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
if (send == SEND_MICROPHONE) {
if (engine()->voe()->base()->StartSend(channel) == -1) {
LOG_RTCERR1(StartSend, channel);
return false;
}
if (engine()->voe()->file() &&
engine()->voe()->file()->StopPlayingFileAsMicrophone(channel) == -1) {
LOG_RTCERR1(StopPlayingFileAsMicrophone, channel);
return false;
}
} else { // SEND_NOTHING
ASSERT(send == SEND_NOTHING);
if (engine()->voe()->base()->StopSend(channel) == -1) {
LOG_RTCERR1(StopSend, channel);
return false;
}
}
return true;
}
void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
if (engine()->voe()->network()->RegisterExternalTransport(
channel, *this) == -1) {
LOG_RTCERR2(RegisterExternalTransport, channel, this);
}
// Enable RTCP (for quality stats and feedback messages)
EnableRtcp(channel);
// Reset all recv codecs; they will be enabled via SetRecvCodecs.
ResetRecvCodecs(channel);
}
bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) {
LOG_RTCERR1(DeRegisterExternalTransport, channel);
}
if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
LOG_RTCERR1(DeleteChannel, channel);
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
// If the default channel is already used for sending create a new channel
// otherwise use the default channel for sending.
int channel = GetSendChannelNum(sp.first_ssrc());
if (channel != -1) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc();
return false;
}
bool default_channel_is_available = true;
for (ChannelMap::const_iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (IsDefaultChannel(iter->second.channel)) {
default_channel_is_available = false;
break;
}
}
if (default_channel_is_available) {
channel = voe_channel();
} else {
// Create a new channel for sending audio data.
channel = engine()->voe()->base()->CreateChannel();
if (channel == -1) {
LOG_RTCERR0(CreateChannel);
return false;
}
ConfigureSendChannel(channel);
}
// Save the channel to send_channels_, so that RemoveSendStream() can still
// delete the channel in case failure happens below.
send_channels_[sp.first_ssrc()] = WebRtcVoiceChannelInfo(channel, NULL);
// Set the send (local) SSRC.
// If there are multiple send SSRCs, we can only set the first one here, and
// the rest of the SSRC(s) need to be set after SetSendCodec has been called
// (with a codec requires multiple SSRC(s)).
if (engine()->voe()->rtp()->SetLocalSSRC(channel, sp.first_ssrc()) == -1) {
LOG_RTCERR2(SetSendSSRC, channel, sp.first_ssrc());
return false;
}
// At this point the channel's local SSRC has been updated. If the channel is
// the default channel make sure that all the receive channels are updated as
// well. Receive channels have to have the same SSRC as the default channel in
// order to send receiver reports with this SSRC.
if (IsDefaultChannel(channel)) {
for (ChannelMap::const_iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
// Only update the SSRC for non-default channels.
if (!IsDefaultChannel(it->second.channel)) {
if (engine()->voe()->rtp()->SetLocalSSRC(it->second.channel,
sp.first_ssrc()) != 0) {
LOG_RTCERR2(SetLocalSSRC, it->second.channel, sp.first_ssrc());
return false;
}
}
}
}
if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
return false;
}
// Set the current codecs to be used for the new channel.
if (!send_codecs_.empty() && !SetSendCodecs(channel, send_codecs_))
return false;
return ChangeSend(channel, desired_send_);
}
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) {
ChannelMap::iterator it = send_channels_.find(ssrc);
if (it == send_channels_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
int channel = it->second.channel;
ChangeSend(channel, SEND_NOTHING);
// Notify the audio renderer that the send channel is going away.
if (it->second.renderer)
it->second.renderer->RemoveChannel(channel);
if (IsDefaultChannel(channel)) {
// Do not delete the default channel since the receive channels depend on
// the default channel, recycle it instead.
ChangeSend(channel, SEND_NOTHING);
} else {
// Clean up and delete the send channel.
LOG(LS_INFO) << "Removing audio send stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
if (!DeleteChannel(channel))
return false;
}
send_channels_.erase(it);
if (send_channels_.empty())
ChangeSend(SEND_NOTHING);
return true;
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
talk_base::CritScope lock(&receive_channels_cs_);
if (!VERIFY(sp.ssrcs.size() == 1))
return false;
uint32 ssrc = sp.first_ssrc();
if (receive_channels_.find(ssrc) != receive_channels_.end()) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
// Reuse default channel for recv stream in non-conference mode call
// when the default channel is not being used.
if (!InConferenceMode() && default_receive_ssrc_ == 0) {
LOG(LS_INFO) << "Recv stream " << sp.first_ssrc()
<< " reuse default channel";
default_receive_ssrc_ = sp.first_ssrc();
receive_channels_.insert(std::make_pair(
default_receive_ssrc_, WebRtcVoiceChannelInfo(voe_channel(), NULL)));
return SetPlayout(voe_channel(), playout_);
}
// 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);
if (!recv_codecs_.empty()) {
for (std::vector<AudioCodec>::const_iterator it = recv_codecs_.begin();
it != recv_codecs_.end(); ++it) {
webrtc::CodecInst voe_codec;
if (engine()->FindWebRtcCodec(*it, &voe_codec)) {
voe_codec.pltype = it->id;
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 (InConferenceMode()) {
// To be in par with the video, voe_channel() is not used for receiving in
// a conference call.
if (receive_channels_.empty() && default_receive_ssrc_ == 0 && 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);
}
}
SetNack(channel, nack_enabled_);
receive_channels_.insert(
std::make_pair(ssrc, WebRtcVoiceChannelInfo(channel, NULL)));
// TODO(juberti): 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::RemoveRecvStream(uint32 ssrc) {
talk_base::CritScope lock(&receive_channels_cs_);
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
if (ssrc == default_receive_ssrc_) {
ASSERT(IsDefaultChannel(it->second.channel));
// Recycle the default channel is for recv stream.
if (playout_)
SetPlayout(voe_channel(), false);
if (it->second.renderer)
it->second.renderer->RemoveChannel(voe_channel());
default_receive_ssrc_ = 0;
receive_channels_.erase(it);
return true;
}
// Non default channel.
// Notify the renderer that channel is going away.
if (it->second.renderer)
it->second.renderer->RemoveChannel(it->second.channel);
LOG(LS_INFO) << "Removing audio stream " << ssrc
<< " with VoiceEngine channel #" << it->second.channel << ".";
if (!DeleteChannel(it->second.channel)) {
// Erase the entry anyhow.
receive_channels_.erase(it);
return false;
}
receive_channels_.erase(it);
bool enable_default_channel_playout = false;
if (receive_channels_.empty()) {
// 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.
// We do this for both conference mode and non-conference mode.
// TODO(oja): Does the default channel still have it's CN state?
enable_default_channel_playout = true;
}
if (!InConferenceMode() && receive_channels_.size() == 1 &&
default_receive_ssrc_ != 0) {
// Only the default channel is active, enable the playout on default
// channel.
enable_default_channel_playout = true;
}
if (enable_default_channel_playout && playout_) {
LOG(LS_INFO) << "Enabling playback on the default voice channel";
SetPlayout(voe_channel(), true);
}
return true;
}
bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc,
AudioRenderer* renderer) {
ChannelMap::iterator it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
if (renderer) {
// Return an error if trying to set a valid renderer with an invalid ssrc.
LOG(LS_ERROR) << "SetRemoteRenderer failed with ssrc "<< ssrc;
return false;
}
// The channel likely has gone away, do nothing.
return true;
}
AudioRenderer* remote_renderer = it->second.renderer;
if (renderer) {
ASSERT(remote_renderer == NULL || remote_renderer == renderer);
if (!remote_renderer) {
renderer->AddChannel(it->second.channel);
}
} else if (remote_renderer) {
// |renderer| == NULL, remove the channel from the renderer.
remote_renderer->RemoveChannel(it->second.channel);
}
// Assign the new value to the struct.
it->second.renderer = renderer;
return true;
}
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc,
AudioRenderer* renderer) {
ChannelMap::iterator it = send_channels_.find(ssrc);
if (it == send_channels_.end()) {
if (renderer) {
// Return an error if trying to set a valid renderer with an invalid ssrc.
LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
return false;
}
// The channel likely has gone away, do nothing.
return true;
}
AudioRenderer* local_renderer = it->second.renderer;
if (renderer) {
ASSERT(local_renderer == NULL || local_renderer == renderer);
if (!local_renderer)
renderer->AddChannel(it->second.channel);
} else if (local_renderer) {
local_renderer->RemoveChannel(it->second.channel);
}
// Assign the new value to the struct.
it->second.renderer = renderer;
return true;
}
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
// In conference mode, the default channel should not be in
// |receive_channels_|.
actives->clear();
for (ChannelMap::iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
int level = GetOutputLevel(it->second.channel);
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 = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
int level = GetOutputLevel(it->second.channel);
highest = talk_base::_max(level, highest);
}
return highest;
}
int WebRtcVoiceMediaChannel::GetTimeSinceLastTyping() {
int ret;
if (engine()->voe()->processing()->TimeSinceLastTyping(ret) == -1) {
// In case of error, log the info and continue
LOG_RTCERR0(TimeSinceLastTyping);
ret = -1;
} else {
ret *= 1000; // We return ms, webrtc returns seconds.
}
return ret;
}
void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
int cost_per_typing, int reporting_threshold, int penalty_decay,
int type_event_delay) {
if (engine()->voe()->processing()->SetTypingDetectionParameters(
time_window, cost_per_typing,
reporting_threshold, penalty_decay, type_event_delay) == -1) {
// In case of error, log the info and continue
LOG_RTCERR5(SetTypingDetectionParameters, time_window,
cost_per_typing, reporting_threshold, penalty_decay,
type_event_delay);
}
}
bool WebRtcVoiceMediaChannel::SetOutputScaling(
uint32 ssrc, double left, double right) {
talk_base::CritScope lock(&receive_channels_cs_);
// Collect the channels to scale the output volume.
std::vector<int> channels;
if (0 == ssrc) { // Collect all channels, including the default one.
// Default channel is not in receive_channels_ if it is not being used for
// playout.
if (default_receive_ssrc_ == 0)
channels.push_back(voe_channel());
for (ChannelMap::const_iterator it = receive_channels_.begin();
it != receive_channels_.end(); ++it) {
channels.push_back(it->second.channel);
}
} else { // Collect only the channel of the specified ssrc.
int channel = GetReceiveChannelNum(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(&receive_channels_cs_);
// Determine which channel based on ssrc.
int channel = (0 == ssrc) ? voe_channel() : GetReceiveChannelNum(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_) {
return false;
}
// The voe file api is not available in chrome.
if (!engine()->voe()->file()) {
return false;
}
// Determine which VoiceEngine channel to play on.
int channel = (ssrc == 0) ? voe_channel() : GetReceiveChannelNum(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::CanInsertDtmf() {
return dtmf_allowed_;
}
bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
int duration, int flags) {
if (!dtmf_allowed_) {
return false;
}
// Send the event.
if (flags & cricket::DF_SEND) {
int channel = -1;
if (ssrc == 0) {
bool default_channel_is_inuse = false;
for (ChannelMap::const_iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
if (IsDefaultChannel(iter->second.channel)) {
default_channel_is_inuse = true;
break;
}