Google Git
Sign in
android / platform / external / webrtc / 68786d20400f1f3744ad83549325665c18ea9e5b / . / talk / media / webrtc / webrtcvoiceengine.cc
blob: bcf1738c85138f355395397983e6887aee1ab54e [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/media/base/audioframe.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/base/base64.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace cricket {
static const int kMaxNumPacketSize = 6;
struct CodecPref {
const char* name;
int clockrate;
int channels;
int payload_type;
bool is_multi_rate;
int packet_sizes_ms[kMaxNumPacketSize];
};
// Note: keep the supported packet sizes in ascending order.
static const CodecPref kCodecPrefs[] = {
{ kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
{ kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
{ kIsacCodecName, 32000, 1, 104, true, { 30 } },
// G722 should be advertised as 8000 Hz because of the RFC "bug".
{ kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
{ kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
{ kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
{ kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
{ kCnCodecName, 32000, 1, 106, false, { } },
{ kCnCodecName, 16000, 1, 105, false, { } },
{ kCnCodecName, 8000, 1, 13, false, { } },
{ kRedCodecName, 8000, 1, 127, false, { } },
{ kDtmfCodecName, 8000, 1, 126, false, { } },
};
// 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;
#else
static const int kDefaultAudioDeviceId = 0;
#endif
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
static const int kNackMaxPackets = 250;
// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
// Recommended bitrates:
// 8-12 kb/s for NB speech,
// 16-20 kb/s for WB speech,
// 28-40 kb/s for FB speech,
// 48-64 kb/s for FB mono music, and
// 64-128 kb/s for FB stereo music.
// The current implementation applies the following values to mono signals,
// and multiplies them by 2 for stereo.
static const int kOpusBitrateNb = 12000;
static const int kOpusBitrateWb = 20000;
static const int kOpusBitrateFb = 32000;
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
static const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
// 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(rtc::LoggingSeverity sev, char* text) {
const char* delim = "\r\n";
for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
LOG_V(sev) << tok;
}
}
// Severity is an integer because it comes is assumed to be from command line.
static int SeverityToFilter(int severity) {
int filter = webrtc::kTraceNone;
switch (severity) {
case rtc::LS_VERBOSE:
filter |= webrtc::kTraceAll;
FALLTHROUGH();
case rtc::LS_INFO:
filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
FALLTHROUGH();
case rtc::LS_WARNING:
filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
FALLTHROUGH();
case rtc::LS_ERROR:
filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
}
return filter;
}
static bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
static bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
return (_stricmp(codec.plname, ref_name) == 0);
}
static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
if (IsCodec(codec, kCodecPrefs[i].name) &&
kCodecPrefs[i].clockrate == codec.plfreq) {
return kCodecPrefs[i].is_multi_rate;
}
}
return false;
}
static bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec) {
for (const AudioCodec& c : codecs) {
if (c.Matches(codec)) {
if (found_codec != NULL) {
*found_codec = c;
}
return true;
}
}
return false;
}
static bool IsNackEnabled(const AudioCodec& codec) {
return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
kParamValueEmpty));
}
static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
for (int packet_size_ms : codec_pref.packet_sizes_ms) {
if (packet_size_ms && packet_size_ms <= ptime_ms) {
selected_packet_size_ms = packet_size_ms;
}
}
return selected_packet_size_ms;
}
// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec
// pacsize if it's valid, or we will pick the next smallest value we support.
// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
for (const CodecPref& codec_pref : kCodecPrefs) {
if ((IsCodec(*codec, codec_pref.name) &&
codec_pref.clockrate == codec->plfreq) ||
IsCodec(*codec, kG722CodecName)) {
int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
if (packet_size_ms) {
// Convert unit from milli-seconds to samples.
codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
return true;
}
}
}
return false;
}
// Return true if codec.params[feature] == "1", false otherwise.
static bool IsCodecFeatureEnabled(const AudioCodec& codec,
const char* feature) {
int value;
return codec.GetParam(feature, &value) && value == 1;
}
// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
// otherwise. If the value (either from params or codec.bitrate) <=0, use the
// default configuration. If the value is beyond feasible bit rate of Opus,
// clamp it. Returns the Opus bit rate for operation.
static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
int bitrate = 0;
bool use_param = true;
if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
bitrate = codec.bitrate;
use_param = false;
}
if (bitrate <= 0) {
if (max_playback_rate <= 8000) {
bitrate = kOpusBitrateNb;
} else if (max_playback_rate <= 16000) {
bitrate = kOpusBitrateWb;
} else {
bitrate = kOpusBitrateFb;
}
if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
bitrate *= 2;
}
} else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
std::string rate_source =
use_param ? "Codec parameter \"maxaveragebitrate\"" :
"Supplied Opus bitrate";
LOG(LS_WARNING) << rate_source
<< " is invalid and is replaced by: "
<< bitrate;
}
return bitrate;
}
// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
int value;
if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
return value;
}
return kOpusDefaultMaxPlaybackRate;
}
static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
bool* enable_codec_fec, int* max_playback_rate,
bool* enable_codec_dtx) {
*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
*enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
*max_playback_rate = GetOpusMaxPlaybackRate(codec);
// 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.
voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
// codec.
static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
if (IsCodec(*voe_codec, kG722CodecName)) {
// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
// has changed, and this special case is no longer needed.
DCHECK(voe_codec->plfreq != new_plfreq);
voe_codec->plfreq = new_plfreq;
}
}
// Gets the default set of options applied to the engine. Historically, these
// were supplied as a combination of flags from the channel manager (ec, agc,
// ns, and highpass) and the rest hardcoded in InitInternal.
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.stereo_swapping.Set(false);
options.audio_jitter_buffer_max_packets.Set(50);
options.audio_jitter_buffer_fast_accelerate.Set(false);
options.typing_detection.Set(true);
options.conference_mode.Set(false);
options.adjust_agc_delta.Set(0);
options.experimental_agc.Set(false);
options.extended_filter_aec.Set(false);
options.delay_agnostic_aec.Set(false);
options.experimental_ns.Set(false);
options.aec_dump.Set(false);
return options;
}
static std::string GetEnableString(bool enable) {
return enable ? "enable" : "disable";
}
WebRtcVoiceEngine::WebRtcVoiceEngine()
: voe_wrapper_(new VoEWrapper()),
tracing_(new VoETraceWrapper()),
adm_(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,
VoETraceWrapper* tracing)
: voe_wrapper_(voe_wrapper),
tracing_(tracing),
adm_(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,
kRtpAudioLevelHeaderExtensionDefaultId));
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
options_ = GetDefaultEngineOptions();
}
void WebRtcVoiceEngine::ConstructCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
webrtc::CodecInst voe_codec;
if (GetVoeCodec(i, &voe_codec)) {
// Skip uncompressed formats.
if (IsCodec(voe_codec, kL16CodecName)) {
continue;
}
const CodecPref* pref = NULL;
for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
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 (IsCodec(codec, kIsacCodecName)) {
// Indicate auto-bitrate in signaling.
codec.bitrate = 0;
}
if (IsCodec(codec, kOpusCodecName)) {
// Only add fmtp parameters that differ from the spec.
if (kPreferredMinPTime != kOpusDefaultMinPTime) {
codec.params[kCodecParamMinPTime] =
rtc::ToString(kPreferredMinPTime);
}
if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
codec.params[kCodecParamMaxPTime] =
rtc::ToString(kPreferredMaxPTime);
}
codec.SetParam(kCodecParamUseInbandFec, 1);
// TODO(hellner): Add ptime, sprop-stereo, and stereo
// 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);
}
bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) {
return false;
}
// Change the sample rate of G722 to 8000 to match SDP.
MaybeFixupG722(codec, 8000);
return true;
}
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;
}
// Test to see if the media processor was deregistered properly
DCHECK(SignalRxMediaFrame.is_empty());
DCHECK(SignalTxMediaFrame.is_empty());
tracing_->SetTraceCallback(NULL);
}
bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
DCHECK(worker_thread == rtc::Thread::Current());
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_filter = log_filter_;
int extended_filter = log_filter_ | SeverityToFilter(rtc::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(rtc::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 (const AudioCodec& codec : codecs_) {
LOG(LS_INFO) << ToString(codec);
}
// Disable the DTMF playout when a tone is sent.
// PlayDtmfTone will be used if local playout is needed.
if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
LOG_RTCERR1(SetDtmfFeedbackStatus, false);
}
initialized_ = true;
return true;
}
void WebRtcVoiceEngine::Terminate() {
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
initialized_ = false;
StopAecDump();
voe_wrapper_->base()->Terminate();
desired_local_monitor_enable_ = false;
}
int WebRtcVoiceEngine::GetCapabilities() {
return AUDIO_SEND | AUDIO_RECV;
}
VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
const AudioOptions& options) {
WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this);
if (!ch->valid()) {
delete ch;
return nullptr;
}
if (!ch->SetOptions(options)) {
LOG(LS_WARNING) << "Failed to set options while creating channel.";
}
return ch;
}
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 (options.aecm_generate_comfort_noise.Get(&aecm_comfort_noise)) {
LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
<< aecm_comfort_noise << " (default is false).";
}
#if defined(IOS)
// On iOS, VPIO provides built-in EC and AGC.
options.echo_cancellation.Set(false);
options.auto_gain_control.Set(false);
LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
#elif defined(ANDROID)
ec_mode = webrtc::kEcAecm;
#endif
#if defined(IOS) || defined(ANDROID)
// Set the AGC mode for iOS as well despite disabling it above, to avoid
// unsupported configuration errors from webrtc.
agc_mode = webrtc::kAgcFixedDigital;
options.typing_detection.Set(false);
options.experimental_agc.Set(false);
options.extended_filter_aec.Set(false);
options.experimental_ns.Set(false);
#endif
// Delay Agnostic AEC automatically turns on EC if not set except on iOS
// where the feature is not supported.
bool use_delay_agnostic_aec = false;
#if !defined(IOS)
if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) {
if (use_delay_agnostic_aec) {
options.echo_cancellation.Set(true);
options.extended_filter_aec.Set(true);
ec_mode = webrtc::kEcConference;
}
}
#endif
LOG(LS_INFO) << "Applying audio options: " << options.ToString();
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
bool echo_cancellation = false;
if (options.echo_cancellation.Get(&echo_cancellation)) {
// Check if platform supports built-in EC. Currently only supported on
// Android and in combination with Java based audio layer.
// TODO(henrika): investigate possibility to support built-in EC also
// in combination with Open SL ES audio.
const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
if (built_in_aec) {
// Built-in EC exists on this device and use_delay_agnostic_aec is not
// overriding it. Enable/Disable it according to the echo_cancellation
// audio option.
const bool enable_built_in_aec =
echo_cancellation && !use_delay_agnostic_aec;
if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
options.echo_cancellation.Set(false);
echo_cancellation = false;
LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
}
}
if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) {
LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode);
return false;
} else {
LOG(LS_INFO) << "Echo control set to " << echo_cancellation
<< " with mode " << ec_mode;
}
#if !defined(ANDROID)
// TODO(ajm): Remove the error return on Android from webrtc.
if (voep->SetEcMetricsStatus(echo_cancellation) == -1) {
LOG_RTCERR1(SetEcMetricsStatus, echo_cancellation);
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;
} else {
LOG(LS_INFO) << "Auto gain set to " << auto_gain_control << " with mode "
<< agc_mode;
}
}
if (options.tx_agc_target_dbov.IsSet() ||
options.tx_agc_digital_compression_gain.IsSet() ||
options.tx_agc_limiter.IsSet()) {
// Override default_agc_config_. Generally, an unset option means "leave
// the VoE bits alone" in this function, so we want whatever is set to be
// stored as the new "default". If we didn't, then setting e.g.
// tx_agc_target_dbov would reset digital compression gain and limiter
// settings.
// Also, if we don't update default_agc_config_, then adjust_agc_delta
// would be an offset from the original values, and not whatever was set
// explicitly.
default_agc_config_.targetLeveldBOv =
options.tx_agc_target_dbov.GetWithDefaultIfUnset(
default_agc_config_.targetLeveldBOv);
default_agc_config_.digitalCompressionGaindB =
options.tx_agc_digital_compression_gain.GetWithDefaultIfUnset(
default_agc_config_.digitalCompressionGaindB);
default_agc_config_.limiterEnable =
options.tx_agc_limiter.GetWithDefaultIfUnset(
default_agc_config_.limiterEnable);
if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
LOG_RTCERR3(SetAgcConfig,
default_agc_config_.targetLeveldBOv,
default_agc_config_.digitalCompressionGaindB,
default_agc_config_.limiterEnable);
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;
} else {
LOG(LS_VERBOSE) << "Noise suppression set to " << noise_suppression
<< " with mode " << ns_mode;
}
}
bool highpass_filter;
if (options.highpass_filter.Get(&highpass_filter)) {
LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
if (voep->EnableHighPassFilter(highpass_filter) == -1) {
LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter);
return false;
}
}
bool stereo_swapping;
if (options.stereo_swapping.Get(&stereo_swapping)) {
LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping;
voep->EnableStereoChannelSwapping(stereo_swapping);
if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) {
LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping);
return false;
}
}
int audio_jitter_buffer_max_packets;
if (options.audio_jitter_buffer_max_packets.Get(
&audio_jitter_buffer_max_packets)) {
LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets;
voe_config_.Set<webrtc::NetEqCapacityConfig>(
new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
}
bool audio_jitter_buffer_fast_accelerate;
if (options.audio_jitter_buffer_fast_accelerate.Get(
&audio_jitter_buffer_fast_accelerate)) {
LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate;
voe_config_.Set<webrtc::NetEqFastAccelerate>(
new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate));
}
bool typing_detection;
if (options.typing_detection.Get(&typing_detection)) {
LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
if (voep->SetTypingDetectionStatus(typing_detection) == -1) {
// 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)) {
LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta;
if (!AdjustAgcLevel(adjust_agc_delta)) {
return false;
}
}
bool aec_dump;
if (options.aec_dump.Get(&aec_dump)) {
LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump;
if (aec_dump)
StartAecDump(kAecDumpByAudioOptionFilename);
else
StopAecDump();
}
webrtc::Config config;
delay_agnostic_aec_.SetFrom(options.delay_agnostic_aec);
bool delay_agnostic_aec;
if (delay_agnostic_aec_.Get(&delay_agnostic_aec)) {
LOG(LS_INFO) << "Delay agnostic aec is enabled? " << delay_agnostic_aec;
config.Set<webrtc::DelayAgnostic>(
new webrtc::DelayAgnostic(delay_agnostic_aec));
}
extended_filter_aec_.SetFrom(options.extended_filter_aec);
bool extended_filter;
if (extended_filter_aec_.Get(&extended_filter)) {
LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter;
config.Set<webrtc::ExtendedFilter>(
new webrtc::ExtendedFilter(extended_filter));
}
experimental_ns_.SetFrom(options.experimental_ns);
bool experimental_ns;
if (experimental_ns_.Get(&experimental_ns)) {
LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
config.Set<webrtc::ExperimentalNs>(
new webrtc::ExperimentalNs(experimental_ns));
}
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
if (audioproc) {
audioproc->SetExtraOptions(config);
}
uint32 recording_sample_rate;
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) {
LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate);
}
}
uint32 playout_sample_rate;
if (options.playout_sample_rate.Get(&playout_sample_rate)) {
LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate;
if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) {
LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate);
}
}
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)
int in_id = in_device ? rtc::FromString<int>(in_device->id) :
kDefaultAudioDeviceId;
int out_id = out_device ? rtc::FromString<int>(out_device->id) :
kDefaultAudioDeviceId;
// The device manager uses -1 as the default device, which was the case for
// VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
#ifndef WIN32
if (-1 == in_id) {
in_id = kDefaultAudioDeviceId;
}
if (-1 == out_id) {
out_id = kDefaultAudioDeviceId;
}
#endif
std::string in_name = (in_id != kDefaultAudioDeviceId) ?
in_device->name : "Default device";
std::string out_name = (out_id != kDefaultAudioDeviceId) ?
out_device->name : "Default device";
LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name
<< ") and speaker to (id=" << out_id << ", name=" << out_name
<< ")";
// 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 (WebRtcVoiceMediaChannel* channel : channels_) {
if (!channel->PausePlayout()) {
LOG(LS_WARNING) << "Failed to pause playout";
ret = false;
}
if (!channel->PauseSend()) {
LOG(LS_WARNING) << "Failed to pause send";
ret = false;
}
}
// Find the recording device id in VoiceEngine and set recording device.
if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) {
ret = false;
}
if (ret) {
if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
ret = false;
}
webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
if (ap)
ap->Initialize();
}
// 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_name, out_id);
ret = false;
}
}
// Resume all audio playback and capture.
for (WebRtcVoiceMediaChannel* channel : channels_) {
if (!channel->ResumePlayout()) {
LOG(LS_WARNING) << "Failed to resume playout";
ret = false;
}
if (!channel->ResumeSend()) {
LOG(LS_WARNING) << "Failed to resume send";
ret = false;
}
}
// 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
}
bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId(
bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) {
// In Linux, VoiceEngine uses the same device dev_id as the device manager.
#if defined(LINUX) || defined(ANDROID)
*rtc_id = dev_id;
return true;
#else
// In Windows and Mac, we need to find the VoiceEngine device id by name
// unless the input dev_id is the default device id.
if (kDefaultAudioDeviceId == dev_id) {
*rtc_id = dev_id;
return true;
}
// Get the number of VoiceEngine audio devices.
int count = 0;
if (is_input) {
if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) {
LOG_RTCERR0(GetNumOfRecordingDevices);
return false;
}
} else {
if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) {
LOG_RTCERR0(GetNumOfPlayoutDevices);
return false;
}
}
for (int i = 0; i < count; ++i) {
char name[128];
char guid[128];
if (is_input) {
voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid);
LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name;
} else {
voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid);
LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name;
}
std::string webrtc_name(name);
if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) {
*rtc_id = i;
return true;
}
}
LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name;
return false;
#endif
}
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) {
DCHECK(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 (GetVoeCodec(i, &voe_codec)) {
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
bool multi_rate = IsCodecMultiRate(voe_codec);
// Allow arbitrary rates for ISAC to be specified.
if (multi_rate) {
// Set codec.bitrate to 0 so the check for codec.Matches() passes.
codec.bitrate = 0;
}
if (codec.Matches(in)) {
if (out) {
// Fixup the payload type.
voe_codec.pltype = in.id;
// Set bitrate if specified.
if (multi_rate && in.bitrate != 0) {
voe_codec.rate = in.bitrate;
}
// Reset G722 sample rate to 16000 to match WebRTC.
MaybeFixupG722(&voe_codec, 16000);
// Apply codec-specific settings.
if (IsCodec(codec, kIsacCodecName)) {
// If ISAC and an explicit bitrate is not specified,
// enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
}
return true;
}
}
}
return false;
}
const std::vector<RtpHeaderExtension>&
WebRtcVoiceEngine::rtp_header_extensions() const {
return rtp_header_extensions_;
}
void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) {
// if min_sev == -1, we keep the current log level.
if (min_sev >= 0) {
SetTraceFilter(SeverityToFilter(min_sev));
}
log_options_ = filter;
SetTraceOptions(initialized_ ? log_options_ : "");
}
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;
rtc::tokenize(options, ' ', '"', '"', &opts);
std::vector<std::string>::iterator tracefile =
std::find(opts.begin(), opts.end(), "tracefile");
if (tracefile != opts.end() && ++tracefile != opts.end()) {
// Write encrypted debug output (at same loglevel) to file
// EncryptedTraceFile no longer supported.
if (tracing_->SetTraceFile(tracefile->c_str()) == -1) {
LOG_RTCERR1(SetTraceFile, *tracefile);
}
}
// Allow trace options to override the trace filter. We default
// it to log_filter_ (as a translation of libjingle log levels)
// elsewhere, but this allows clients to explicitly set webrtc
// log levels.
std::vector<std::string>::iterator tracefilter =
std::find(opts.begin(), opts.end(), "tracefilter");
if (tracefilter != opts.end() && ++tracefilter != opts.end()) {
if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) {
LOG_RTCERR1(SetTraceFilter, *tracefilter);
}
}
// Set AEC dump file
std::vector<std::string>::iterator recordEC =
std::find(opts.begin(), opts.end(), "recordEC");
if (recordEC != opts.end()) {
++recordEC;
if (recordEC != opts.end())
StartAecDump(recordEC->c_str());
else
StopAecDump();
}
}
// 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) {
rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
sev = rtc::LS_ERROR;
else if (level == webrtc::kTraceWarning)
sev = rtc::LS_WARNING;
else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
sev = rtc::LS_INFO;
else if (level == webrtc::kTraceTerseInfo)
sev = rtc::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) {
rtc::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)) {
DCHECK(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 {
DCHECK(channel != NULL && ssrc != NULL);
*channel = NULL;
*ssrc = 0;
// Find corresponding channel and ssrc
for (WebRtcVoiceMediaChannel* ch : channels_) {
DCHECK(ch != NULL);
if (ch->FindSsrc(channel_num, ssrc)) {
*channel = ch;
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) {
DCHECK(channel_num != NULL);
DCHECK(direction == MPD_RX || direction == MPD_TX);
*channel_num = -1;
// Find corresponding channel for ssrc.
for (const WebRtcVoiceMediaChannel* ch : channels_) {
DCHECK(ch != NULL);
if (direction & MPD_RX) {
*channel_num = ch->GetReceiveChannelNum(ssrc);
}
if (*channel_num == -1 && (direction & MPD_TX)) {
*channel_num = ch->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) {
rtc::CritScope lock(&channels_cs_);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
rtc::CritScope lock(&channels_cs_);
ChannelList::iterator i = std::find(channels_.begin(),
channels_.end(),
channel);
if (i != channels_.end()) {
channels_.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) {
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();
}
return true;
}
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
if (!aec_dump_file_stream) {
LOG(LS_ERROR) << "Could not open AEC dump file stream.";
if (!rtc::ClosePlatformFile(file))
LOG(LS_WARNING) << "Could not close file.";
return false;
}
StopAecDump();
if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StartDebugRecording);
fclose(aec_dump_file_stream);
return false;
}
is_dumping_aec_ = true;
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;
{
rtc::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;
{
rtc::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[],
size_t length,
int sampling_freq,
bool is_stereo) {
rtc::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_RTCERR1(StartDebugRecording, filename.c_str());
} 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;
}
}
int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) {
return voice_engine_wrapper->base()->CreateChannel(voe_config_);
}
int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
return CreateVoiceChannel(voe_wrapper_.get());
}
class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
: public AudioRenderer::Sink {
public:
WebRtcVoiceChannelRenderer(int ch,
webrtc::AudioTransport* voe_audio_transport)
: channel_(ch),
voe_audio_transport_(voe_audio_transport),
renderer_(NULL) {}
~WebRtcVoiceChannelRenderer() override { Stop(); }
// Starts the rendering by setting a sink to the renderer to get data
// callback.
// This method is called on the libjingle worker thread.
// TODO(xians): Make sure Start() is called only once.
void Start(AudioRenderer* renderer) {
rtc::CritScope lock(&lock_);
DCHECK(renderer != NULL);
if (renderer_ != NULL) {
DCHECK(renderer_ == renderer);
return;
}
// TODO(xians): Remove AddChannel() call after Chrome turns on APM
// in getUserMedia by default.
renderer->AddChannel(channel_);
renderer->SetSink(this);
renderer_ = renderer;
}
// Stops rendering by setting the sink of the renderer to NULL. No data
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void Stop() {
rtc::CritScope lock(&lock_);
if (renderer_ == NULL)
return;
renderer_->RemoveChannel(channel_);
renderer_->SetSink(NULL);
renderer_ = NULL;
}
// AudioRenderer::Sink implementation.
// This method is called on the audio thread.
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) override {
voe_audio_transport_->OnData(channel_,
audio_data,
bits_per_sample,
sample_rate,
number_of_channels,
number_of_frames);
}
// Callback from the |renderer_| when it is going away. In case Start() has
// never been called, this callback won't be triggered.
void OnClose() override {
rtc::CritScope lock(&lock_);
// Set |renderer_| to NULL to make sure no more callback will get into
// the renderer.
renderer_ = NULL;
}
// Accessor to the VoE channel ID.
int channel() const { return channel_; }
private:
const int channel_;
webrtc::AudioTransport* const voe_audio_transport_;
// Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
// PeerConnection will make sure invalidating the pointer before the object
// goes away.
AudioRenderer* renderer_;
// Protects |renderer_| in Start(), Stop() and OnClose().
rtc::CriticalSection lock_;
};
// WebRtcVoiceMediaChannel
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine)
: engine_(engine),
voe_channel_(engine->CreateMediaVoiceChannel()),
send_bitrate_setting_(false),
send_bitrate_bps_(0),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
nack_enabled_(false),
playout_(false),
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
call_(nullptr),
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();
DCHECK(receive_streams_.empty() || call_);
// 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);
}
DCHECK(receive_streams_.empty());
// Delete the default channel.
DeleteChannel(voe_channel());
}
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
return (SetSendCodecs(params.codecs) &&
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps) &&
SetOptions(params.options));
}
bool WebRtcVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
// Check if DSCP value is changed from previous.
bool dscp_option_changed = (options_.dscp != options.dscp);
// 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.
}
// Receiver-side auto gain control happens per channel, so set it here from
// options. Note that, like conference mode, setting it on the engine won't
// have the desired effect, since voice channels don't inherit options from
// the media engine when those options are applied per-channel.
bool rx_auto_gain_control;
if (options.rx_auto_gain_control.Get(&rx_auto_gain_control)) {
if (engine()->voe()->processing()->SetRxAgcStatus(
voe_channel(), rx_auto_gain_control,
webrtc::kAgcFixedDigital) == -1) {
LOG_RTCERR1(SetRxAgcStatus, rx_auto_gain_control);
return false;
} else {
LOG(LS_VERBOSE) << "Rx auto gain set to " << rx_auto_gain_control
<< " with mode " << webrtc::kAgcFixedDigital;
}
}
if (options.rx_agc_target_dbov.IsSet() ||
options.rx_agc_digital_compression_gain.IsSet() ||
options.rx_agc_limiter.IsSet()) {
webrtc::AgcConfig config;
// If only some of the options are being overridden, get the current
// settings for the channel and bail if they aren't available.
if (!options.rx_agc_target_dbov.IsSet() ||
!options.rx_agc_digital_compression_gain.IsSet() ||
!options.rx_agc_limiter.IsSet()) {
if (engine()->voe()->processing()->GetRxAgcConfig(
voe_channel(), config) != 0) {
LOG(LS_ERROR) << "Failed to get default rx agc configuration for "
<< "channel " << voe_channel() << ". Since not all rx "
<< "agc options are specified, unable to safely set rx "
<< "agc options.";
return false;
}
}
config.targetLeveldBOv =
options.rx_agc_target_dbov.GetWithDefaultIfUnset(
config.targetLeveldBOv);
config.digitalCompressionGaindB =
options.rx_agc_digital_compression_gain.GetWithDefaultIfUnset(
config.digitalCompressionGaindB);
config.limiterEnable = options.rx_agc_limiter.GetWithDefaultIfUnset(
config.limiterEnable);
if (engine()->voe()->processing()->SetRxAgcConfig(
voe_channel(), config) == -1) {
LOG_RTCERR4(SetRxAgcConfig, voe_channel(), config.targetLeveldBOv,
config.digitalCompressionGaindB, config.limiterEnable);
return false;
}
}
if (dscp_option_changed) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
if (options_.dscp.GetWithDefaultIfUnset(false))
dscp = kAudioDscpValue;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
}
}
SetCall(call_);
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 (const AudioCodec& codec : codecs) {
AudioCodec old_codec;
if (FindCodec(recv_codecs_, codec, &old_codec)) {
if (old_codec.id != codec.id) {
LOG(LS_ERROR) << codec.name << " payload type changed.";
return false;
}
} else {
new_codecs.push_back(codec);
}
}
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 result = SetRecvCodecsInternal(new_codecs);
if (result) {
recv_codecs_ = codecs;
}
if (desired_playout_ && !playout_) {
ResumePlayout();
}
return result;
}
bool WebRtcVoiceMediaChannel::SetSendCodecs(
int channel, const std::vector<AudioCodec>& codecs) {
// Disable VAD, FEC, and RED unless we know the other side wants them.
engine()->voe()->codec()->SetVADStatus(channel, false);
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
engine()->voe()->rtp()->SetREDStatus(channel, false);
engine()->voe()->codec()->SetFECStatus(channel, false);
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD and DTMF.
bool found_send_codec = false;
webrtc::CodecInst send_codec;
memset(&send_codec, 0, sizeof(send_codec));
bool nack_enabled = nack_enabled_;
bool enable_codec_fec = false;
bool enable_opus_dtx = false;
int opus_max_playback_rate = 0;
// Set send codec (the first non-telephone-event/CN codec)
for (const AudioCodec& codec : codecs) {
// Ignore codecs we don't know about. The negotiation step should prevent
// this, but double-check to be sure.
webrtc::CodecInst voe_codec;
if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
// Skip telephone-event/CN codec, which will be handled later.
continue;
}
// 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 RED audio, is a special case where the actual codec to be
// used is specified in params.
if (IsCodec(codec, kRedCodecName)) {
// Parse out the RED parameters. If we fail, just ignore RED;
// we don't support all possible params/usage scenarios.
if (!GetRedSendCodec(codec, codecs, &send_codec)) {
continue;
}
// Enable redundant encoding of the specified codec. Treat any
// failure as a fatal internal error.
LOG(LS_INFO) << "Enabling RED on channel " << channel;
if (engine()->voe()->rtp()->SetREDStatus(channel, true, codec.id) == -1) {
LOG_RTCERR3(SetREDStatus, channel, true, codec.id);
return false;
}
} else {
send_codec = voe_codec;
nack_enabled = IsNackEnabled(codec);
// For Opus as the send codec, we are to determine inband FEC, maximum
// playback rate, and opus internal dtx.
if (IsCodec(codec, kOpusCodecName)) {
GetOpusConfig(codec, &send_codec, &enable_codec_fec,
&opus_max_playback_rate, &enable_opus_dtx);
}
// Set packet size if the AudioCodec param kCodecParamPTime is set.
int ptime_ms = 0;
if (codec.GetParam(kCodecParamPTime, &ptime_ms)) {
if (!SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
LOG(LS_WARNING) << "Failed to set packet size for codec "
<< send_codec.plname;
return false;
}
}
}
found_send_codec = true;
break;
}
if (nack_enabled_ != nack_enabled) {
SetNack(channel, nack_enabled);
nack_enabled_ = nack_enabled;
}
if (!found_send_codec) {
LOG(LS_WARNING) << "Received empty list of codecs.";
return false;
}
// Set the codec immediately, since SetVADStatus() depends on whether
// the current codec is mono or stereo.
if (!SetSendCodec(channel, send_codec))
return false;
// FEC should be enabled after SetSendCodec.
if (enable_codec_fec) {
LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
<< channel;
if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
// Enable codec internal FEC. Treat any failure as fatal internal error.
LOG_RTCERR2(SetFECStatus, channel, true);
return false;
}
}
if (IsCodec(send_codec, kOpusCodecName)) {
// DTX and maxplaybackrate should be set after SetSendCodec. Because current
// send codec has to be Opus.
// Set Opus internal DTX.
LOG(LS_INFO) << "Attempt to "
<< GetEnableString(enable_opus_dtx)
<< " Opus DTX on channel "
<< channel;
if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) {
LOG_RTCERR2(SetOpusDtx, channel, enable_opus_dtx);
return false;
}
// If opus_max_playback_rate <= 0, the default maximum playback rate
// (48 kHz) will be used.
if (opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
<< opus_max_playback_rate
<< " Hz on channel "
<< channel;
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
channel, opus_max_playback_rate) == -1) {
LOG_RTCERR2(SetOpusMaxPlaybackRate, channel, opus_max_playback_rate);
return false;
}
}
}
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
if (send_bitrate_setting_) {
SetSendBitrateInternal(send_bitrate_bps_);
}
// Loop through the codecs list again to config the telephone-event/CN codec.
for (const AudioCodec& codec : codecs) {
// Ignore codecs we don't know about. The negotiation step should prevent
// this, but double-check to be sure.
webrtc::CodecInst voe_codec;
if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
// Find the DTMF telephone event "codec" and tell VoiceEngine channels
// about it.
if (IsCodec(codec, kDtmfCodecName)) {
if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
channel, codec.id) == -1) {
LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, codec.id);
return false;
}
} else if (IsCodec(codec, kCnCodecName)) {
// Turn voice activity detection/comfort noise on if supported.
// Set the wideband CN payload type appropriately.
// (narrowband always uses the static payload type 13).
webrtc::PayloadFrequencies cn_freq;
switch (codec.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 " << codec.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, codec.id, cn_freq) == -1) {
LOG_RTCERR3(SetSendCNPayloadType, channel, codec.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 (codec.clockrate == send_codec.plfreq && send_codec.channels != 2) {
// TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
// interaction between VAD and Opus FEC.
LOG(LS_INFO) << "Enabling VAD";
if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
LOG_RTCERR2(SetVADStatus, channel, true);
return false;
}
}
}
}
return true;
}
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
dtmf_allowed_ = false;
for (const AudioCodec& codec : codecs) {
// Find the DTMF telephone event "codec".
if (IsCodec(codec, kDtmfCodecName)) {
dtmf_allowed_ = true;
}
}
// Cache the codecs in order to configure the channel created later.
send_codecs_ = codecs;
for (const auto& ch : send_channels_) {
if (!SetSendCodecs(ch.second->channel(), codecs)) {
return false;
}
}
// Set nack status on receive channels and update |nack_enabled_|.
SetNack(receive_channels_, nack_enabled_);
return true;
}
void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels,
bool nack_enabled) {
for (const auto& ch : channels) {
SetNack(ch.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 (const auto& ch : send_channels_) {
if (!SetSendCodec(ch.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;
webrtc::CodecInst current_codec;
if (engine()->voe()->codec()->GetSendCodec(channel, current_codec) == 0 &&
(send_codec == current_codec)) {
// Codec is already configured, we can return without setting it again.
return true;
}
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) {
if (receive_extensions_ == extensions) {
return true;
}
// The default channel may or may not be in |receive_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelRecvRtpHeaderExtensions(voe_channel(), extensions)) {
return false;
}
// Loop through all receive channels and enable/disable the extensions.
for (const auto& ch : receive_channels_) {
if (!SetChannelRecvRtpHeaderExtensions(ch.second->channel(), extensions)) {
return false;
}
}
receive_extensions_ = extensions;
// Recreate AudioReceiveStream:s.
{
std::vector<webrtc::RtpExtension> exts;
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (audio_level_extension) {
exts.push_back({
kRtpAudioLevelHeaderExtension, audio_level_extension->id});
}
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (send_time_extension) {
exts.push_back({
kRtpAbsoluteSenderTimeHeaderExtension, send_time_extension->id});
}
recv_rtp_extensions_.swap(exts);
SetCall(call_);
}
return true;
}
bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id,
audio_level_extension)) {
return false;
}
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
if (send_extensions_ == extensions) {
return true;
}
// The default channel may or may not be in |send_channels_|. Set the rtp
// header extensions for default channel regardless.
if (!SetChannelSendRtpHeaderExtensions(voe_channel(), extensions)) {
return false;
}
// Loop through all send channels and enable/disable the extensions.
for (const auto& ch : send_channels_) {
if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
return false;
}
}
send_extensions_ = extensions;
return true;
}
bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions(
int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
const RtpHeaderExtension* audio_level_extension =
FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
audio_level_extension)) {
return false;
}
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
if (!SetHeaderExtension(
&webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
send_time_extension)) {
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
desired_playout_ = playout;
return ChangePlayout(desired_playout_);
}
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 (const auto& ch : receive_channels_) {
if (!SetPlayout(ch.second->channel(), playout)) {
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
<< ch.second->channel() << " failed";
result = false;
break;
}
}
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 (const auto& ch : send_channels_) {
if (!ChangeSend(ch.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
DCHECK(send == SEND_NOTHING);
if (engine()->voe()->base()->StopSend(channel) == -1) {
LOG_RTCERR1(StopSend, channel);
return false;
}
}
return true;
}
// TODO(ronghuawu): Change this method to return bool.
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);
// Set RTP header extension for the new channel.
SetChannelSendRtpHeaderExtensions(channel, send_extensions_);
}
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 (const auto& ch : send_channels_) {
if (IsDefaultChannel(ch.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()->CreateMediaVoiceChannel();
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.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
send_channels_.insert(
std::make_pair(sp.first_ssrc(),
new WebRtcVoiceChannelRenderer(channel, audio_transport)));
// 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 (const auto& ch : receive_channels_) {
// Only update the SSRC for non-default channels.
if (!IsDefaultChannel(ch.second->channel())) {
if (engine()->voe()->rtp()->SetLocalSSRC(ch.second->channel(),
sp.first_ssrc()) != 0) {
LOG_RTCERR2(SetLocalSSRC, ch.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);
// Delete the WebRtcVoiceChannelRenderer object connected to the channel,
// this will disconnect the audio renderer with the send channel.
delete it->second;
send_channels_.erase(it);
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;
}
if (send_channels_.empty())
ChangeSend(SEND_NOTHING);
return true;
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
DCHECK(thread_checker_.CalledOnValidThread());
rtc::CritScope lock(&receive_channels_cs_);
if (!VERIFY(sp.ssrcs.size() == 1))
return false;
uint32 ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
return false;
}
if (receive_channels_.find(ssrc) != receive_channels_.end()) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end());
// Reuse default channel for recv stream in non-conference mode call
// when the default channel is not being used.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
if (!InConferenceMode() && default_receive_ssrc_ == 0) {
LOG(LS_INFO) << "Recv stream " << ssrc << " reuse default channel";
default_receive_ssrc_ = ssrc;
WebRtcVoiceChannelRenderer* channel_renderer =
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport);
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
receive_stream_params_[ssrc] = sp;
TryAddAudioRecvStream(ssrc);
return SetPlayout(voe_channel(), playout_);
}
// Create a new channel for receiving audio data.
int channel = engine()->CreateMediaVoiceChannel();
if (channel == -1) {
LOG_RTCERR0(CreateChannel);
return false;
}
if (!ConfigureRecvChannel(channel)) {
DeleteChannel(channel);
return false;
}
WebRtcVoiceChannelRenderer* channel_renderer =
new WebRtcVoiceChannelRenderer(channel, audio_transport);
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
receive_stream_params_[ssrc] = sp;
TryAddAudioRecvStream(ssrc);
LOG(LS_INFO) << "New audio stream " << ssrc
<< " registered to VoiceEngine channel #"
<< channel << ".";
return true;
}
bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
// 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 = 0;
webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp();
if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) {
LOG_RTCERR1(GetSendSSRC, channel);
return false;
}
if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) {
LOG_RTCERR1(SetSendSSRC, channel);
return false;
}
// Associate receive channel to default channel (so the receive channel can
// obtain RTT from the send channel)
engine()->voe()->base()->AssociateSendChannel(channel, voe_channel());
LOG(LS_INFO) << "VoiceEngine channel #"
<< channel << " is associated with channel #"
<< voe_channel() << ".";
// Use the same recv payload types as our default channel.
ResetRecvCodecs(channel);
if (!recv_codecs_.empty()) {
for (const auto& codec : recv_codecs_) {
webrtc::CodecInst voe_codec;
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
voe_codec.pltype = codec.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_);
// Set RTP header extension for the new channel.
if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) {
return false;
}
return SetPlayout(channel, playout_);
}
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) {
DCHECK(thread_checker_.CalledOnValidThread());
rtc::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;
}
TryRemoveAudioRecvStream(ssrc);
receive_stream_params_.erase(ssrc);
// Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
// will disconnect the audio renderer with the receive channel.
// Cache the channel before the deletion.
const int channel = it->second->channel();
delete it->second;
receive_channels_.erase(it);
if (ssrc == default_receive_ssrc_) {
DCHECK(IsDefaultChannel(channel));
// Recycle the default channel is for recv stream.
if (playout_)
SetPlayout(voe_channel(), false);
default_receive_ssrc_ = 0;
return true;
}
LOG(LS_INFO) << "Removing audio stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
if (!DeleteChannel(channel))
return false;
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;
}
if (renderer)
it->second->Start(renderer);
else
it->second->Stop();
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;
}
if (renderer)
it->second->Start(renderer);
else
it->second->Stop();
return true;
}
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
// In conference mode, the default channel should not be in
// |receive_channels_|.
actives->clear();
for (const auto& ch : receive_channels_) {
int level = GetOutputLevel(ch.second->channel());
if (level > 0) {
actives->push_back(std::make_pair(ch.first, level));
}
}
return true;
}
int WebRtcVoiceMediaChannel::GetOutputLevel() {
// return the highest output level of all streams
int highest = GetOutputLevel(voe_channel());
for (const auto& ch : receive_channels_) {
int level = GetOutputLevel(ch.second->channel());
highest = std::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) {
rtc::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 (const auto& ch : receive_channels_) {
channels.push_back(ch.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>(std::max(left, right));
if (scale > 0.0001f) {
left /= scale;
right /= scale;
}
for (int ch_id : channels) {
if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(
ch_id, scale)) {
LOG_RTCERR2(SetChannelOutputVolumeScaling, ch_id, scale);
return false;
}
if (-1 == engine()->voe()->volume()->SetOutputVolumePan(
ch_id, static_cast<float>(left), static_cast<float>(right))) {
LOG_RTCERR3(SetOutputVolumePan, ch_id, 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 " << ch_id << " and ssrc " << ssrc;
}
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 (const auto& ch : send_channels_) {
if (IsDefaultChannel(ch.second->channel())) {
default_channel_is_inuse = true;
break;
}
}
if (default_channel_is_inuse) {
channel = voe_channel();
} else if (!send_channels_.empty()) {
channel = send_channels_.begin()->second->channel();
}
} else {
channel = GetSendChannelNum(ssrc);
}
if (channel == -1) {
LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
<< ssrc << " is not in use.";
return false;
}
// Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
if (engine()->voe()->dtmf()->SendTelephoneEvent(
channel, event, true, duration) == -1) {
LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration);
return false;
}
}
// Play the event.
if (flags & cricket::DF_PLAY) {
// Play DTMF tone locally.
if (engine()->voe()->dtmf()->PlayDtmfTone(event, duration) == -1) {
LOG_RTCERR2(PlayDtmfTone, event, duration);
return false;
}
}
return true;
}
void WebRtcVoiceMediaChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
DCHECK(thread_checker_.CalledOnValidThread());
// If hooked up to a "Call", forward packet there too.
if (call_) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
call_->Receiver()->DeliverPacket(
webrtc::MediaType::AUDIO,
reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
webrtc_packet_time);
}
// 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 =
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), 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_.
if (engine()->voe()->file()) {
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->size(),
webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
DCHECK(thread_checker_.CalledOnValidThread());
// If hooked up to a "Call", forward packet there too.
if (call_) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
call_->Receiver()->DeliverPacket(
webrtc::MediaType::AUDIO,
reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
webrtc_packet_time);
}
// Sending channels need all RTCP packets with feedback information.
// Even sender reports can contain attached report blocks.
// Receiving channels need sender reports in order to create
// correct receiver reports.
int type = 0;
if (!GetRtcpType(packet->data(), packet->size(), &type)) {
LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
return;
}
// If it is a sender report, find the channel that is listening.
bool has_sent_to_default_channel = false;
if (type == kRtcpTypeSR) {
int which_channel =
GetReceiveChannelNum(ParseSsrc(packet->data(), packet->size(), true));
if (which_channel != -1) {
engine()->voe()->network()->ReceivedRTCPPacket(
which_channel, packet->data(), packet->size());
if (IsDefaultChannel(which_channel))
has_sent_to_default_channel = true;
}
}
// SR may continue RR and any RR entry may correspond to any one of the send
// channels. So all RTCP packets must be forwarded all send channels. VoE
// will filter out RR internally.
for (const auto& ch : send_channels_) {
// Make sure not sending the same packet to default channel more than once.
if (IsDefaultChannel(ch.second->channel()) &&
has_sent_to_default_channel)
continue;
engine()->voe()->network()->ReceivedRTCPPacket(
ch.second->channel(), packet->data(), packet->size());
}
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
LOG_RTCERR2(SetInputMute, channel, muted);
return false;
}
// We set the AGC to mute state only when all the channels are muted.
// This implementation is not ideal, instead we should signal the AGC when
// the mic channel is muted/unmuted. We can't do it today because there
// is no good way to know which stream is mapping to the mic channel.
bool all_muted = muted;
for (const auto& ch : send_channels_) {
if (!all_muted) {
break;
}
if (engine()->voe()->volume()->GetInputMute(ch.second->channel(),
all_muted)) {
LOG_RTCERR1(GetInputMute, ch.second->channel());
return false;
}
}
webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
if (ap)
ap->set_output_will_be_muted(all_muted);
return true;
}
// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
// SetMaxSendBitrate() in future.
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
return SetSendBitrateInternal(bps);
}
bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
send_bitrate_setting_ = true;
send_bitrate_bps_ = bps;
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet. "
<< "The send bitrate setting will be applied later.";
return true;
}
// Bitrate is auto by default.
// TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
// SetMaxSendBandwith(0), the second call removes the previous limit.
if (bps <= 0)
return true;
webrtc::CodecInst codec = *send_codec_;
bool is_multi_rate = IsCodecMultiRate(codec);
if (is_multi_rate) {
// If codec is multi-rate then just set the bitrate.
codec.rate = bps;
if (!SetSendCodec(codec)) {
LOG(LS_INFO) << "Failed to set codec " << codec.plname
<< " to bitrate " << bps << " bps.";
return false;
}
return true;
} else {
// If codec is not multi-rate and |bps| is less than the fixed bitrate
// then fail. If codec is not multi-rate and |bps| exceeds or equal the
// fixed bitrate then ignore.
if (bps < codec.rate) {
LOG(LS_INFO) << "Failed to set codec " << codec.plname
<< " to bitrate " << bps << " bps"
<< ", requires at least " << codec.rate << " bps.";
return false;
}
return true;
}
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
bool echo_metrics_on = false;
// These can take on valid negative values, so use the lowest possible level
// as default rather than -1.
int echo_return_loss = -100;
int 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.
int echo_delay_median_ms = -1;
int echo_delay_std_ms = -1;
if (engine()->voe()->processing()->GetEcMetricsStatus(
echo_metrics_on) != -1 && echo_metrics_on) {
// TODO(ajm): 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) {
echo_return_loss = erl;
echo_return_loss_enhancement = erle;
}
int median, std;
float dummy;
if (engine()->voe()->processing()->GetEcDelayMetrics(
median, std, dummy) != -1) {
echo_delay_median_ms = median;
echo_delay_std_ms = std;
}
}
webrtc::CallStatistics cs;
unsigned int ssrc;
webrtc::CodecInst codec;
unsigned int level;
for (const auto& ch : send_channels_) {
const int channel = ch.second->channel();
// 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;
if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
continue;
}
sinfo.add_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;
// Get data from the last remote RTCP report. Use default values if no data
// available.
sinfo.fraction_lost = -1.0;
sinfo.jitter_ms = -1;
sinfo.packets_lost = -1;
sinfo.ext_seqnum = -1;
std::vector<webrtc::ReportBlock> receive_blocks;
if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
channel, &receive_blocks) != -1 &&
engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
for (const webrtc::ReportBlock& block : receive_blocks) {
// Lookup report for send ssrc only.
if (block.source_SSRC == sinfo.ssrc()) {
// Convert Q8 to floating point.
sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256;
// Convert samples to milliseconds.
if (codec.plfreq / 1000 > 0) {
sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000);
}
sinfo.packets_lost = block.cumulative_num_packets_lost;
sinfo.ext_seqnum = block.extended_highest_sequence_number;
break;
}
}
}
// Local speech level.
sinfo.audio_level = (engine()->voe()->volume()->
GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
// TODO(xians): We are injecting the same APM logging to all the send
// channels here because there is no good way to know which send channel
// is using the APM. The correct fix is to allow the send channels to have
// their own APM so that we can feed the correct APM logging to different
// send channels. See issue crbug/264611 .
sinfo.echo_return_loss = echo_return_loss;
sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
sinfo.echo_delay_median_ms = echo_delay_median_ms;
sinfo.echo_delay_std_ms = echo_delay_std_ms;
// TODO(ajm): Re-enable this metric once we have a reliable implementation.
sinfo.aec_quality_min = -1;
sinfo.typing_noise_detected = typing_noise_detected_;
info->senders.push_back(sinfo);
}
// Build the list of receivers, one for each receiving channel, or 1 in
// a 1:1 call.
std::vector<int> channels;
for (const auto& ch : receive_channels_) {
channels.push_back(ch.second->channel());
}
if (channels.empty()) {
channels.push_back(voe_channel());
}
// Get the SSRC and stats for each receiver, based on our own calculations.
for (int ch_id : channels) {
memset(&cs, 0, sizeof(cs));
if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 &&
engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 &&
engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) {
VoiceReceiverInfo rinfo;
rinfo.add_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;
rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
if (codec.pltype != -1) {
rinfo.codec_name = codec.plname;
}
// 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(
ch_id, ns) != -1) {
rinfo.jitter_buffer_ms = ns.currentBufferSize;
rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
rinfo.expand_rate =
static_cast<float>(ns.currentExpandRate) / (1 << 14);
rinfo.speech_expand_rate =
static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
rinfo.secondary_decoded_rate =
static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
rinfo.accelerate_rate =
static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
rinfo.preemptive_expand_rate =
static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
}
webrtc::AudioDecodingCallStats ds;
if (engine()->voe()->neteq() &&
engine()->voe()->neteq()->GetDecodingCallStatistics(
ch_id, &ds) != -1) {
rinfo.decoding_calls_to_silence_generator =
ds.calls_to_silence_generator;
rinfo.decoding_calls_to_neteq = ds.calls_to_neteq;
rinfo.decoding_normal = ds.decoded_normal;
rinfo.decoding_plc = ds.decoded_plc;
rinfo.decoding_cng = ds.decoded_cng;
rinfo.decoding_plc_cng = ds.decoded_plc_cng;
}
if (engine()->voe()->sync()) {
int jitter_buffer_delay_ms = 0;
int playout_buffer_delay_ms = 0;
engine()->voe()->sync()->GetDelayEstimate(
ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms);
rinfo.delay_estimate_ms = jitter_buffer_delay_ms +
playout_buffer_delay_ms;
}
// Get speech level.
rinfo.audio_level = (engine()->voe()->volume()->
GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1;
info->receivers.push_back(rinfo);
}
}
return true;
}
void WebRtcVoiceMediaChannel::GetLastMediaError(
uint32* ssrc, VoiceMediaChannel::Error* error) {
DCHECK(ssrc != NULL);
DCHECK(error != NULL);
FindSsrc(voe_channel(), ssrc);
*error = WebRtcErrorToChannelError(GetLastEngineError());
}
bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
rtc::CritScope lock(&receive_channels_cs_);
DCHECK(ssrc != NULL);
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 sending channel.
for (const auto& ch : send_channels_) {
if (ch.second->channel() == channel_num) {
// This is a sending channel.
uint32 local_ssrc = 0;
if (engine()->voe()->rtp()->GetLocalSSRC(
channel_num, local_ssrc) != -1) {
*ssrc = local_ssrc;
}
return true;
}
}
// Check whether this is a receiving channel.
for (const auto& ch : receive_channels_) {
if (ch.second->channel() == channel_num) {
*ssrc = ch.first;
return true;
}
}
}
return false;
}
void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
if (error == VE_TYPING_NOISE_WARNING) {
typing_noise_detected_ = true;
} else if (error == VE_TYPING_NOISE_OFF_WARNING) {
typing_noise_detected_ = false;
}
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::GetReceiveChannelNum(uint32 ssrc) const {
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
if (it != receive_channels_.end())
return it->second->channel();
return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
}
int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) const {
ChannelMap::const_iterator it = send_channels_.find(ssrc);
if (it != send_channels_.end())
return it->second->channel();
return -1;
}
void WebRtcVoiceMediaChannel::SetCall(webrtc::Call* call) {
DCHECK(thread_checker_.CalledOnValidThread());
for (const auto& it : receive_channels_) {
TryRemoveAudioRecvStream(it.first);
}
call_ = call;
for (const auto& it : receive_channels_) {
TryAddAudioRecvStream(it.first);
}
}
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 (rtc::split(red_params, '/', &red_pts) != 2 ||
red_pts[0] != red_pts[1] ||
!rtc::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|.
for (const AudioCodec& codec : all_codecs) {
if (codec.id == red_pt) {
// If we find the right codec, that will be the codec we pass to
// SetSendCodec, with the desired payload type.
if (engine()->FindWebRtcCodec(codec, send_codec)) {
return true;
} else {
break;
}
}
}
LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
return false;
}
bool WebRtcVoiceMediaChannel::EnableRtcp(int channel) {
if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
LOG_RTCERR2(SetRTCPStatus, channel, 1);
return false;
}
// TODO(juberti): 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 = rtc::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;
}
}
bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
int channel_id, const RtpHeaderExtension* extension) {
bool enable = false;
int id = 0;
std::string uri;
if (extension) {
enable = true;
id = extension->id;
uri = extension->uri;
}
if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
LOG_RTCERR4(*setter, uri, channel_id, enable, id);
return false;
}
return true;
}
void WebRtcVoiceMediaChannel::TryAddAudioRecvStream(uint32 ssrc) {
DCHECK(thread_checker_.CalledOnValidThread());
WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
DCHECK(channel != nullptr);
DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
// If we are hooked up to a webrtc::Call, create an AudioReceiveStream too.
if (!call_) {
return;
}
webrtc::AudioReceiveStream::Config config;
config.rtp.remote_ssrc = ssrc;
// Only add RTP extensions if we support combined A/V BWE.
config.rtp.extensions = recv_rtp_extensions_;
config.combined_audio_video_bwe =
options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
config.voe_channel_id = channel->channel();
config.sync_group = receive_stream_params_[ssrc].sync_label;
webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
receive_streams_.insert(std::make_pair(ssrc, s));
}
void WebRtcVoiceMediaChannel::TryRemoveAudioRecvStream(uint32 ssrc) {
DCHECK(thread_checker_.CalledOnValidThread());
// If we are hooked up to a webrtc::Call, assume there is an
// AudioReceiveStream to destroy too.
if (call_) {
auto stream_it = receive_streams_.find(ssrc);
if (stream_it != receive_streams_.end()) {
call_->DestroyAudioReceiveStream(stream_it->second);
receive_streams_.erase(stream_it);
}
}
}
bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
const std::vector<AudioCodec>& new_codecs) {
for (const AudioCodec& codec : new_codecs) {
webrtc::CodecInst voe_codec;
if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
LOG(LS_INFO) << ToString(codec);
voe_codec.pltype = codec.id;
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));
return false;
}
}
// Set the receive codecs on all receiving channels.
for (const auto& ch : receive_channels_) {
if (engine()->voe()->codec()->SetRecPayloadType(
ch.second->channel(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
ToString(voe_codec));
return false;
}
}
} else {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
return false;
}
}
return true;
}
int WebRtcSoundclipStream::Read(void *buf, size_t len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
return static_cast<int>(res);
}
int WebRtcSoundclipStream::Rewind() {
mem_.Rewind();
// Return -1 to keep VoiceEngine from looping.
return (loop_) ? 0 : -1;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE