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webrtc / src / webrtc / 8262976b30119307af82f875c3ad7821c5d94d7f / . / media / engine / webrtcvoiceengine.cc
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/*
* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifdef HAVE_WEBRTC_VOICE
#include "webrtc/media/engine/webrtcvoiceengine.h"
#include <algorithm>
#include <cstdio>
#include <functional>
#include <string>
#include <vector>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/race_checker.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/engine/payload_type_mapper.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
#include "webrtc/media/engine/webrtcvoe.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace cricket {
namespace {
const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
webrtc::kTraceWarning | webrtc::kTraceError |
webrtc::kTraceCritical;
const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
webrtc::kTraceInfo;
// 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.
#ifdef WIN32
const int kDefaultAudioDeviceId = -1;
#elif !defined(WEBRTC_IOS)
const int kDefaultAudioDeviceId = 0;
#endif
constexpr int kNackRtpHistoryMs = 5000;
// Check to verify that the define for the intelligibility enhancer is properly
// set.
#if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \
(WEBRTC_INTELLIGIBILITY_ENHANCER != 0 && \
WEBRTC_INTELLIGIBILITY_ENHANCER != 1)
#error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1"
#endif
// 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.
const int kOpusBitrateNbBps = 12000;
const int kOpusBitrateWbBps = 20000;
const int kOpusBitrateFbBps = 32000;
// Opus bitrate should be in the range between 6000 and 510000.
const int kOpusMinBitrateBps = 6000;
const int kOpusMaxBitrateBps = 510000;
// iSAC bitrate should be <= 56000.
const int kIsacMaxBitrateBps = 56000;
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
// Constants from voice_engine_defines.h.
const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
const int kMaxTelephoneEventCode = 255;
const int kMinTelephoneEventDuration = 100;
const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16
const int kMinPayloadType = 0;
const int kMaxPayloadType = 127;
class ProxySink : public webrtc::AudioSinkInterface {
public:
ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }
void OnData(const Data& audio) override { sink_->OnData(audio); }
private:
webrtc::AudioSinkInterface* sink_;
};
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
return false;
}
if (sp.ssrcs.size() > 1) {
LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
return false;
}
return true;
}
// Dumps an AudioCodec in RFC 2327-ish format.
std::string ToString(const AudioCodec& codec) {
std::stringstream ss;
ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
<< " (" << codec.id << ")";
return ss.str();
}
std::string ToString(const webrtc::CodecInst& codec) {
std::stringstream ss;
ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
<< " (" << codec.pltype << ")";
return ss.str();
}
bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
return (_stricmp(codec.plname, ref_name) == 0);
}
bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec) {
for (const AudioCodec& c : codecs) {
if (c.Matches(codec)) {
if (found_codec != NULL) {
*found_codec = c;
}
return true;
}
}
return false;
}
bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
if (codecs.empty()) {
return true;
}
std::vector<int> payload_types;
for (const AudioCodec& codec : codecs) {
payload_types.push_back(codec.id);
}
std::sort(payload_types.begin(), payload_types.end());
auto it = std::unique(payload_types.begin(), payload_types.end());
return it == payload_types.end();
}
// Return true if codec.params[feature] == "1", false otherwise.
bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) {
int value;
return codec.GetParam(feature, &value) && value == 1;
}
rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
const AudioOptions& options) {
if (options.audio_network_adaptor && *options.audio_network_adaptor &&
options.audio_network_adaptor_config) {
// Turn on audio network adaptor only when |options_.audio_network_adaptor|
// equals true and |options_.audio_network_adaptor_config| has a value.
return options.audio_network_adaptor_config;
}
return rtc::Optional<std::string>();
}
// Returns integer parameter params[feature] if it is defined. Returns
// |default_value| otherwise.
int GetCodecFeatureInt(const AudioCodec& codec,
const char* feature,
int default_value) {
int value = 0;
if (codec.GetParam(feature, &value)) {
return value;
}
return default_value;
}
// 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.
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 = kOpusBitrateNbBps;
} else if (max_playback_rate <= 16000) {
bitrate = kOpusBitrateWbBps;
} else {
bitrate = kOpusBitrateFbBps;
}
if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
bitrate *= 2;
}
} else if (bitrate < kOpusMinBitrateBps || bitrate > kOpusMaxBitrateBps) {
bitrate = (bitrate < kOpusMinBitrateBps) ? kOpusMinBitrateBps
: kOpusMaxBitrateBps;
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;
}
void GetOpusConfig(const AudioCodec& codec,
webrtc::CodecInst* voe_codec,
bool* enable_codec_fec,
int* max_playback_rate,
bool* enable_codec_dtx,
int* min_ptime_ms,
int* max_ptime_ms) {
*enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
*enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
*max_playback_rate = GetCodecFeatureInt(codec, kCodecParamMaxPlaybackRate,
kOpusDefaultMaxPlaybackRate);
*max_ptime_ms =
GetCodecFeatureInt(codec, kCodecParamMaxPTime, kOpusDefaultMaxPTime);
*min_ptime_ms =
GetCodecFeatureInt(codec, kCodecParamMinPTime, kOpusDefaultMinPTime);
if (*max_ptime_ms < *min_ptime_ms) {
// If min ptime or max ptime defined by codec parameter is wrong, we use
// the default values.
*max_ptime_ms = kOpusDefaultMaxPTime;
*min_ptime_ms = kOpusDefaultMinPTime;
}
// 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);
}
webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
webrtc::AudioState::Config config;
config.voice_engine = voe_wrapper->engine();
return config;
}
class WebRtcVoiceCodecs final {
public:
// TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
// list and add a test which verifies VoE supports the listed codecs.
static std::vector<AudioCodec> SupportedSendCodecs() {
std::vector<AudioCodec> result;
// Iterate first over our preferred codecs list, so that the results are
// added in order of preference.
for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
const CodecPref* pref = &kCodecPrefs[i];
for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
// Change the sample rate of G722 to 8000 to match SDP.
MaybeFixupG722(&voe_codec, 8000);
// Skip uncompressed formats.
if (IsCodec(voe_codec, kL16CodecName)) {
continue;
}
if (!IsCodec(voe_codec, pref->name) ||
pref->clockrate != voe_codec.plfreq ||
pref->channels != voe_codec.channels) {
// Not a match.
continue;
}
AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels);
LOG(LS_INFO) << "Adding supported codec: " << 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);
codec.AddFeedbackParam(
FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
// TODO(hellner): Add ptime, sprop-stereo, and stereo
// when they can be set to values other than the default.
}
result.push_back(codec);
}
}
return result;
}
static bool ToCodecInst(const AudioCodec& in,
webrtc::CodecInst* out) {
for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
// Change the sample rate of G722 to 8000 to match SDP.
MaybeFixupG722(&voe_codec, 8000);
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels);
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;
}
static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
if (IsCodec(codec, kCodecPrefs[i].name) &&
kCodecPrefs[i].clockrate == codec.plfreq) {
return kCodecPrefs[i].is_multi_rate;
}
}
return false;
}
static int MaxBitrateBps(const webrtc::CodecInst& codec) {
for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
if (IsCodec(codec, kCodecPrefs[i].name) &&
kCodecPrefs[i].clockrate == codec.plfreq) {
return kCodecPrefs[i].max_bitrate_bps;
}
}
return 0;
}
// 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;
}
static const AudioCodec* GetPreferredCodec(
const std::vector<AudioCodec>& codecs,
webrtc::CodecInst* out) {
RTC_DCHECK(out);
// Select the preferred send codec (the first non-telephone-event/CN codec).
for (const AudioCodec& codec : codecs) {
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.
// Ignore codecs we don't know about. The negotiation step should prevent
// this, but double-check to be sure.
if (!ToCodecInst(codec, out)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
return &codec;
}
return nullptr;
}
private:
static const int kMaxNumPacketSize = 6;
struct CodecPref {
const char* name;
int clockrate;
size_t channels;
int payload_type;
bool is_multi_rate;
int packet_sizes_ms[kMaxNumPacketSize];
int max_bitrate_bps;
};
// Note: keep the supported packet sizes in ascending order.
static const CodecPref kCodecPrefs[11];
static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
for (int packet_size_ms : codec_pref.packet_sizes_ms) {
if (packet_size_ms && packet_size_ms <= ptime_ms) {
selected_packet_size_ms = packet_size_ms;
}
}
return selected_packet_size_ms;
}
// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
// codec.
static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
if (IsCodec(*voe_codec, kG722CodecName)) {
// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
// has changed, and this special case is no longer needed.
RTC_DCHECK(voe_codec->plfreq != new_plfreq);
voe_codec->plfreq = new_plfreq;
}
}
};
const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[11] = {
{kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
{kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
{kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
// 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, {}},
{kDtmfCodecName, 8000, 1, 126, false, {}}};
rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
int rtp_max_bitrate_bps,
const webrtc::CodecInst& codec_inst) {
const int bps = MinPositive(max_send_bitrate_bps, rtp_max_bitrate_bps);
const int codec_rate = codec_inst.rate;
if (bps <= 0) {
return rtc::Optional<int>(codec_rate);
}
if (codec_inst.pltype == -1) {
return rtc::Optional<int>(codec_rate);
;
}
if (WebRtcVoiceCodecs::IsCodecMultiRate(codec_inst)) {
// If codec is multi-rate then just set the bitrate.
return rtc::Optional<int>(
std::min(bps, WebRtcVoiceCodecs::MaxBitrateBps(codec_inst)));
}
if (bps < codec_inst.rate) {
// 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.
LOG(LS_ERROR) << "Failed to set codec " << codec_inst.plname
<< " to bitrate " << bps << " bps"
<< ", requires at least " << codec_inst.rate << " bps.";
return rtc::Optional<int>();
}
return rtc::Optional<int>(codec_rate);
}
} // namespace {
bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
webrtc::CodecInst* out) {
return WebRtcVoiceCodecs::ToCodecInst(in, out);
}
WebRtcVoiceEngine::WebRtcVoiceEngine(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
: WebRtcVoiceEngine(adm, decoder_factory, new VoEWrapper()) {
audio_state_ = webrtc::AudioState::Create(MakeAudioStateConfig(voe()));
}
WebRtcVoiceEngine::WebRtcVoiceEngine(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
VoEWrapper* voe_wrapper)
: adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
RTC_DCHECK(voe_wrapper);
RTC_DCHECK(decoder_factory);
signal_thread_checker_.DetachFromThread();
// Load our audio codec list.
LOG(LS_INFO) << "Supported send codecs in order of preference:";
send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
for (const AudioCodec& codec : send_codecs_) {
LOG(LS_INFO) << ToString(codec);
}
LOG(LS_INFO) << "Supported recv codecs in order of preference:";
recv_codecs_ = CollectRecvCodecs();
for (const AudioCodec& codec : recv_codecs_) {
LOG(LS_INFO) << ToString(codec);
}
channel_config_.enable_voice_pacing = true;
// Temporarily turn logging level up for the Init() call.
webrtc::Trace::SetTraceCallback(this);
webrtc::Trace::set_level_filter(kElevatedTraceFilter);
LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
decoder_factory_));
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
// No ADM supplied? Get the default one from VoE.
if (!adm_) {
adm_ = voe_wrapper_->base()->audio_device_module();
}
RTC_DCHECK(adm_);
apm_ = voe_wrapper_->base()->audio_processing();
RTC_DCHECK(apm_);
// Save the default AGC configuration settings. This must happen before
// calling ApplyOptions or the default will be overwritten.
int error = voe_wrapper_->processing()->GetAgcConfig(default_agc_config_);
RTC_DCHECK_EQ(0, error);
// Set default engine options.
{
AudioOptions options;
options.echo_cancellation = rtc::Optional<bool>(true);
options.auto_gain_control = rtc::Optional<bool>(true);
options.noise_suppression = rtc::Optional<bool>(true);
options.highpass_filter = rtc::Optional<bool>(true);
options.stereo_swapping = rtc::Optional<bool>(false);
options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
options.typing_detection = rtc::Optional<bool>(true);
options.adjust_agc_delta = rtc::Optional<int>(0);
options.experimental_agc = rtc::Optional<bool>(false);
options.extended_filter_aec = rtc::Optional<bool>(false);
options.delay_agnostic_aec = rtc::Optional<bool>(false);
options.experimental_ns = rtc::Optional<bool>(false);
options.intelligibility_enhancer = rtc::Optional<bool>(false);
options.level_control = rtc::Optional<bool>(false);
bool error = ApplyOptions(options);
RTC_DCHECK(error);
}
SetDefaultDevices();
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
StopAecDump();
voe_wrapper_->base()->Terminate();
webrtc::Trace::SetTraceCallback(nullptr);
}
rtc::scoped_refptr<webrtc::AudioState>
WebRtcVoiceEngine::GetAudioState() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return audio_state_;
}
VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return new WebRtcVoiceMediaChannel(this, config, options, call);
}
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: " << options_in.ToString();
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;
if (options.aecm_generate_comfort_noise) {
LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
<< *options.aecm_generate_comfort_noise
<< " (default is false).";
}
#if defined(WEBRTC_IOS)
// On iOS, VPIO provides built-in EC, NS and AGC.
options.echo_cancellation = rtc::Optional<bool>(false);
options.auto_gain_control = rtc::Optional<bool>(false);
options.noise_suppression = rtc::Optional<bool>(false);
LOG(LS_INFO)
<< "Always disable AEC, NS and AGC on iOS. Use built-in instead.";
#elif defined(ANDROID)
ec_mode = webrtc::kEcAecm;
#endif
#if defined(WEBRTC_IOS) || defined(ANDROID)
// Set the AGC mode for iOS as well despite disabling it above, to avoid
// unsupported configuration errors from webrtc.
agc_mode = webrtc::kAgcFixedDigital;
options.typing_detection = rtc::Optional<bool>(false);
options.experimental_agc = rtc::Optional<bool>(false);
options.extended_filter_aec = rtc::Optional<bool>(false);
options.experimental_ns = rtc::Optional<bool>(false);
#endif
// Delay Agnostic AEC automatically turns on EC if not set except on iOS
// where the feature is not supported.
bool use_delay_agnostic_aec = false;
#if !defined(WEBRTC_IOS)
if (options.delay_agnostic_aec) {
use_delay_agnostic_aec = *options.delay_agnostic_aec;
if (use_delay_agnostic_aec) {
options.echo_cancellation = rtc::Optional<bool>(true);
options.extended_filter_aec = rtc::Optional<bool>(true);
ec_mode = webrtc::kEcConference;
}
}
#endif
#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 0)
// Hardcode the intelligibility enhancer to be off.
options.intelligibility_enhancer = rtc::Optional<bool>(false);
#endif
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
if (options.echo_cancellation) {
// Check if platform supports built-in EC. Currently only supported on
// Android and in combination with Java based audio layer.
// TODO(henrika): investigate possibility to support built-in EC also
// in combination with Open SL ES audio.
const bool built_in_aec = adm()->BuiltInAECIsAvailable();
if (built_in_aec) {
// Built-in EC exists on this device and use_delay_agnostic_aec is not
// overriding it. Enable/Disable it according to the echo_cancellation
// audio option.
const bool enable_built_in_aec =
*options.echo_cancellation && !use_delay_agnostic_aec;
if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
options.echo_cancellation = rtc::Optional<bool>(false);
LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
}
}
if (voep->SetEcStatus(*options.echo_cancellation, ec_mode) == -1) {
LOG_RTCERR2(SetEcStatus, *options.echo_cancellation, ec_mode);
return false;
} else {
LOG(LS_INFO) << "Echo control set to " << *options.echo_cancellation
<< " with mode " << ec_mode;
}
#if !defined(ANDROID)
// TODO(ajm): Remove the error return on Android from webrtc.
if (voep->SetEcMetricsStatus(*options.echo_cancellation) == -1) {
LOG_RTCERR1(SetEcMetricsStatus, *options.echo_cancellation);
return false;
}
#endif
if (ec_mode == webrtc::kEcAecm) {
bool cn = options.aecm_generate_comfort_noise.value_or(false);
if (voep->SetAecmMode(aecm_mode, cn) != 0) {
LOG_RTCERR2(SetAecmMode, aecm_mode, cn);
return false;
}
}
}
if (options.auto_gain_control) {
bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();
if (built_in_agc_avaliable) {
if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
*options.auto_gain_control) {
// Disable internal software AGC if built-in AGC is enabled,
// i.e., replace the software AGC with the built-in AGC.
options.auto_gain_control = rtc::Optional<bool>(false);
LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
}
}
if (voep->SetAgcStatus(*options.auto_gain_control, agc_mode) == -1) {
LOG_RTCERR2(SetAgcStatus, *options.auto_gain_control, agc_mode);
return false;
} else {
LOG(LS_INFO) << "Auto gain set to " << *options.auto_gain_control
<< " with mode " << agc_mode;
}
}
if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
options.tx_agc_limiter) {
// Override default_agc_config_. Generally, an unset option means "leave
// the VoE bits alone" in this function, so we want whatever is set to be
// stored as the new "default". If we didn't, then setting e.g.
// tx_agc_target_dbov would reset digital compression gain and limiter
// settings.
// 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.value_or(
default_agc_config_.targetLeveldBOv);
default_agc_config_.digitalCompressionGaindB =
options.tx_agc_digital_compression_gain.value_or(
default_agc_config_.digitalCompressionGaindB);
default_agc_config_.limiterEnable =
options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);
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;
}
}
if (options.intelligibility_enhancer) {
intelligibility_enhancer_ = options.intelligibility_enhancer;
}
if (intelligibility_enhancer_ && *intelligibility_enhancer_) {
LOG(LS_INFO) << "Enabling NS when Intelligibility Enhancer is active.";
options.noise_suppression = intelligibility_enhancer_;
}
if (options.noise_suppression) {
if (adm()->BuiltInNSIsAvailable()) {
bool builtin_ns =
*options.noise_suppression &&
!(intelligibility_enhancer_ && *intelligibility_enhancer_);
if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
// Disable internal software NS if built-in NS is enabled,
// i.e., replace the software NS with the built-in NS.
options.noise_suppression = rtc::Optional<bool>(false);
LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
}
}
if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) {
LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode);
return false;
} else {
LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression
<< " with mode " << ns_mode;
}
}
if (options.highpass_filter) {
LOG(LS_INFO) << "High pass filter enabled? " << *options.highpass_filter;
if (voep->EnableHighPassFilter(*options.highpass_filter) == -1) {
LOG_RTCERR1(SetHighpassFilterStatus, *options.highpass_filter);
return false;
}
}
if (options.stereo_swapping) {
LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
voep->EnableStereoChannelSwapping(*options.stereo_swapping);
if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) {
LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping);
return false;
}
}
if (options.audio_jitter_buffer_max_packets) {
LOG(LS_INFO) << "NetEq capacity is "
<< *options.audio_jitter_buffer_max_packets;
channel_config_.acm_config.neteq_config.max_packets_in_buffer =
std::max(20, *options.audio_jitter_buffer_max_packets);
}
if (options.audio_jitter_buffer_fast_accelerate) {
LOG(LS_INFO) << "NetEq fast mode? "
<< *options.audio_jitter_buffer_fast_accelerate;
channel_config_.acm_config.neteq_config.enable_fast_accelerate =
*options.audio_jitter_buffer_fast_accelerate;
}
if (options.typing_detection) {
LOG(LS_INFO) << "Typing detection is enabled? "
<< *options.typing_detection;
if (voep->SetTypingDetectionStatus(*options.typing_detection) == -1) {
// In case of error, log the info and continue
LOG_RTCERR1(SetTypingDetectionStatus, *options.typing_detection);
}
}
if (options.adjust_agc_delta) {
LOG(LS_INFO) << "Adjust agc delta is " << *options.adjust_agc_delta;
if (!AdjustAgcLevel(*options.adjust_agc_delta)) {
return false;
}
}
webrtc::Config config;
if (options.delay_agnostic_aec)
delay_agnostic_aec_ = options.delay_agnostic_aec;
if (delay_agnostic_aec_) {
LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_;
config.Set<webrtc::DelayAgnostic>(
new webrtc::DelayAgnostic(*delay_agnostic_aec_));
}
if (options.extended_filter_aec) {
extended_filter_aec_ = options.extended_filter_aec;
}
if (extended_filter_aec_) {
LOG(LS_INFO) << "Extended filter aec is enabled? " << *extended_filter_aec_;
config.Set<webrtc::ExtendedFilter>(
new webrtc::ExtendedFilter(*extended_filter_aec_));
}
if (options.experimental_ns) {
experimental_ns_ = options.experimental_ns;
}
if (experimental_ns_) {
LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
config.Set<webrtc::ExperimentalNs>(
new webrtc::ExperimentalNs(*experimental_ns_));
}
if (intelligibility_enhancer_) {
LOG(LS_INFO) << "Intelligibility Enhancer is enabled? "
<< *intelligibility_enhancer_;
config.Set<webrtc::Intelligibility>(
new webrtc::Intelligibility(*intelligibility_enhancer_));
}
if (options.level_control) {
level_control_ = options.level_control;
}
LOG(LS_INFO) << "Level control: "
<< (!!level_control_ ? *level_control_ : -1);
webrtc::AudioProcessing::Config apm_config;
if (level_control_) {
apm_config.level_controller.enabled = *level_control_;
if (options.level_control_initial_peak_level_dbfs) {
apm_config.level_controller.initial_peak_level_dbfs =
*options.level_control_initial_peak_level_dbfs;
}
}
apm()->SetExtraOptions(config);
apm()->ApplyConfig(apm_config);
if (options.recording_sample_rate) {
LOG(LS_INFO) << "Recording sample rate is "
<< *options.recording_sample_rate;
if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
}
}
if (options.playout_sample_rate) {
LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
}
}
return true;
}
void WebRtcVoiceEngine::SetDefaultDevices() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
#if !defined(WEBRTC_IOS)
int in_id = kDefaultAudioDeviceId;
int out_id = kDefaultAudioDeviceId;
LOG(LS_INFO) << "Setting microphone to (id=" << in_id
<< ") and speaker to (id=" << out_id << ")";
bool ret = true;
if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
LOG_RTCERR1(SetRecordingDevice, in_id);
ret = false;
}
apm()->Initialize();
if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
LOG_RTCERR1(SetPlayoutDevice, out_id);
ret = false;
}
if (ret) {
LOG(LS_INFO) << "Set microphone to (id=" << in_id
<< ") and speaker to (id=" << out_id << ")";
}
#endif // !WEBRTC_IOS
}
int WebRtcVoiceEngine::GetInputLevel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
unsigned int ulevel;
return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
static_cast<int>(ulevel) : -1;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return send_codecs_;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return recv_codecs_;
}
RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAudioLevelDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
"Enabled") {
capabilities.header_extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
}
return capabilities;
}
int WebRtcVoiceEngine::GetLastEngineError() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->error();
}
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
// Note: This callback can happen on any thread!
rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
sev = rtc::LS_ERROR;
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);
LOG_V(sev) << "webrtc: " << msg;
}
}
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(channel);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = std::find(channels_.begin(), channels_.end(), channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
// Adjusts the default AGC target level by the specified delta.
// NB: If we start messing with other config fields, we'll want
// to save the current webrtc::AgcConfig as well.
bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::AgcConfig config = default_agc_config_;
config.targetLeveldBOv -= delta;
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::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
if (!aec_dump_file_stream) {
LOG(LS_ERROR) << "Could not open AEC dump file stream.";
if (!rtc::ClosePlatformFile(file))
LOG(LS_WARNING) << "Could not close file.";
return false;
}
StopAecDump();
if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StartDebugRecording);
fclose(aec_dump_file_stream);
return false;
}
is_dumping_aec_ = true;
return true;
}
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!is_dumping_aec_) {
// Start dumping AEC when we are not dumping.
if (apm()->StartDebugRecording(filename.c_str(), -1) !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR1(StartDebugRecording, filename.c_str());
} else {
is_dumping_aec_ = true;
}
}
}
void WebRtcVoiceEngine::StopAecDump() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (is_dumping_aec_) {
// Stop dumping AEC when we are dumping.
if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StopDebugRecording);
}
is_dumping_aec_ = false;
}
}
int WebRtcVoiceEngine::CreateVoEChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->base()->CreateChannel(channel_config_);
}
webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(adm_);
return adm_;
}
webrtc::AudioProcessing* WebRtcVoiceEngine::apm() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(apm_);
return apm_;
}
AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
PayloadTypeMapper mapper;
AudioCodecs out;
const std::vector<webrtc::AudioCodecSpec>& specs =
decoder_factory_->GetSupportedDecoders();
// Only generate CN payload types for these clockrates
std::map<int, bool, std::greater<int>> generate_cn = {{ 8000, false },
{ 16000, false },
{ 32000, false }};
auto map_format = [&mapper, &out] (const webrtc::SdpAudioFormat& format) {
rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
if (!opt_codec) {
LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
return false;
}
auto& codec = *opt_codec;
if (IsCodec(codec, kOpusCodecName)) {
// TODO(ossu): Set this specifically for Opus for now, until we have a
// better way of dealing with rtcp-fb parameters.
codec.AddFeedbackParam(
FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
}
out.push_back(codec);
return true;
};
for (const auto& spec : specs) {
if (map_format(spec.format) && spec.allow_comfort_noise) {
// Generate a CN entry if the decoder allows it and we support the
// clockrate.
auto cn = generate_cn.find(spec.format.clockrate_hz);
if (cn != generate_cn.end()) {
cn->second = true;
}
}
}
// Add CN codecs after "proper" audio codecs
for (const auto& cn : generate_cn) {
if (cn.second) {
map_format({kCnCodecName, cn.first, 1});
}
}
// Add telephone-event codec last
map_format({kDtmfCodecName, 8000, 1});
return out;
}
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioSource::Sink {
public:
WebRtcAudioSendStream(
int ch,
webrtc::AudioTransport* voe_audio_transport,
uint32_t ssrc,
const std::string& c_name,
const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
const std::vector<webrtc::RtpExtension>& extensions,
int max_send_bitrate_bps,
const rtc::Optional<std::string>& audio_network_adaptor_config,
webrtc::Call* call,
webrtc::Transport* send_transport)
: voe_audio_transport_(voe_audio_transport),
call_(call),
config_(send_transport),
max_send_bitrate_bps_(max_send_bitrate_bps),
rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
RTC_DCHECK_GE(ch, 0);
// TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
// RTC_DCHECK(voe_audio_transport);
RTC_DCHECK(call);
config_.rtp.ssrc = ssrc;
config_.rtp.c_name = c_name;
config_.voe_channel_id = ch;
config_.rtp.extensions = extensions;
config_.audio_network_adaptor_config = audio_network_adaptor_config;
RecreateAudioSendStream(send_codec_spec);
}
~WebRtcAudioSendStream() override {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
ClearSource();
call_->DestroyAudioSendStream(stream_);
}
void RecreateAudioSendStream(
const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
send_codec_spec_ = send_codec_spec;
config_.rtp.nack.rtp_history_ms =
send_codec_spec_.nack_enabled ? kNackRtpHistoryMs : 0;
config_.send_codec_spec = send_codec_spec_;
auto send_rate = ComputeSendBitrate(
max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps,
send_codec_spec.codec_inst);
if (send_rate) {
// Apply a send rate that abides by |max_send_bitrate_bps_| and
// |rtp_parameters_| when possible. Otherwise use the codec rate.
config_.send_codec_spec.codec_inst.rate = *send_rate;
}
RecreateAudioSendStream();
}
void RecreateAudioSendStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
config_.rtp.extensions = extensions;
RecreateAudioSendStream();
}
void RecreateAudioSendStream(
const rtc::Optional<std::string>& audio_network_adaptor_config) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (config_.audio_network_adaptor_config == audio_network_adaptor_config) {
return;
}
config_.audio_network_adaptor_config = audio_network_adaptor_config;
RecreateAudioSendStream();
}
bool SetMaxSendBitrate(int bps) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto send_rate =
ComputeSendBitrate(bps, rtp_parameters_.encodings[0].max_bitrate_bps,
send_codec_spec_.codec_inst);
if (!send_rate) {
return false;
}
max_send_bitrate_bps_ = bps;
if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
// Recreate AudioSendStream with new bit rate.
config_.send_codec_spec.codec_inst.rate = *send_rate;
RecreateAudioSendStream();
}
return true;
}
bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
}
void SetSend(bool send) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
send_ = send;
UpdateSendState();
}
void SetMuted(bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
stream_->SetMuted(muted);
muted_ = muted;
}
bool muted() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return muted_;
}
webrtc::AudioSendStream::Stats GetStats() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetStats();
}
// Starts the sending by setting ourselves as a sink to the AudioSource to
// get data callbacks.
// This method is called on the libjingle worker thread.
// TODO(xians): Make sure Start() is called only once.
void SetSource(AudioSource* source) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(source);
if (source_) {
RTC_DCHECK(source_ == source);
return;
}
source->SetSink(this);
source_ = source;
UpdateSendState();
}
// Stops sending by setting the sink of the AudioSource to nullptr. No data
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void ClearSource() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (source_) {
source_->SetSink(nullptr);
source_ = nullptr;
}
UpdateSendState();
}
// AudioSource::Sink implementation.
// This method is called on the audio thread.
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
RTC_DCHECK(voe_audio_transport_);
voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,
bits_per_sample, sample_rate,
number_of_channels, number_of_frames);
}
// Callback from the |source_| when it is going away. In case Start() has
// never been called, this callback won't be triggered.
void OnClose() override {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set |source_| to nullptr to make sure no more callback will get into
// the source.
source_ = nullptr;
UpdateSendState();
}
// Accessor to the VoE channel ID.
int channel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_.voe_channel_id;
}
const webrtc::RtpParameters& rtp_parameters() const {
return rtp_parameters_;
}
bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
RTC_CHECK_EQ(1UL, parameters.encodings.size());
auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
parameters.encodings[0].max_bitrate_bps,
send_codec_spec_.codec_inst);
if (!send_rate) {
return false;
}
rtp_parameters_ = parameters;
// parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
// Recreate AudioSendStream with new bit rate.
config_.send_codec_spec.codec_inst.rate = *send_rate;
RecreateAudioSendStream();
} else {
// parameters.encodings[0].active could have changed.
UpdateSendState();
}
return true;
}
private:
void UpdateSendState() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
stream_->Start();
} else { // !send || source_ = nullptr
stream_->Stop();
}
}
void RecreateAudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioSendStream(stream_);
stream_ = nullptr;
}
RTC_DCHECK(!stream_);
if (webrtc::field_trial::FindFullName("WebRTC-AdaptAudioBitrate") ==
"Enabled") {
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
config_.min_bitrate_bps = kOpusMinBitrateBps;
config_.max_bitrate_bps = kOpusBitrateFbBps;
}
stream_ = call_->CreateAudioSendStream(config_);
RTC_CHECK(stream_);
UpdateSendState();
}
rtc::ThreadChecker worker_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
webrtc::Call* call_ = nullptr;
webrtc::AudioSendStream::Config config_;
// The stream is owned by WebRtcAudioSendStream and may be reallocated if
// configuration changes.
webrtc::AudioSendStream* stream_ = nullptr;
// Raw pointer to AudioSource owned by LocalAudioTrackHandler.
// PeerConnection will make sure invalidating the pointer before the object
// goes away.
AudioSource* source_ = nullptr;
bool send_ = false;
bool muted_ = false;
int max_send_bitrate_bps_;
webrtc::RtpParameters rtp_parameters_;
webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};
class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
public:
WebRtcAudioReceiveStream(
int ch,
uint32_t remote_ssrc,
uint32_t local_ssrc,
bool use_transport_cc,
bool use_nack,
const std::string& sync_group,
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call,
webrtc::Transport* rtcp_send_transport,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
: call_(call), config_() {
RTC_DCHECK_GE(ch, 0);
RTC_DCHECK(call);
config_.rtp.remote_ssrc = remote_ssrc;
config_.rtcp_send_transport = rtcp_send_transport;
config_.voe_channel_id = ch;
config_.sync_group = sync_group;
config_.decoder_factory = decoder_factory;
RecreateAudioReceiveStream(local_ssrc,
use_transport_cc,
use_nack,
extensions);
}
~WebRtcAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
call_->DestroyAudioReceiveStream(stream_);
}
void RecreateAudioReceiveStream(uint32_t local_ssrc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RecreateAudioReceiveStream(local_ssrc,
config_.rtp.transport_cc,
config_.rtp.nack.rtp_history_ms != 0,
config_.rtp.extensions);
}
void RecreateAudioReceiveStream(bool use_transport_cc, bool use_nack) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RecreateAudioReceiveStream(config_.rtp.local_ssrc,
use_transport_cc,
use_nack,
config_.rtp.extensions);
}
void RecreateAudioReceiveStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RecreateAudioReceiveStream(config_.rtp.local_ssrc,
config_.rtp.transport_cc,
config_.rtp.nack.rtp_history_ms != 0,
extensions);
}
webrtc::AudioReceiveStream::Stats GetStats() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetStats();
}
int channel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_.voe_channel_id;
}
void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
stream_->SetSink(std::move(sink));
}
void SetOutputVolume(double volume) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
stream_->SetGain(volume);
}
void SetPlayout(bool playout) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
if (playout) {
LOG(LS_INFO) << "Starting playout for channel #" << channel();
stream_->Start();
} else {
LOG(LS_INFO) << "Stopping playout for channel #" << channel();
stream_->Stop();
}
playout_ = playout;
}
private:
void RecreateAudioReceiveStream(
uint32_t local_ssrc,
bool use_transport_cc,
bool use_nack,
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioReceiveStream(stream_);
stream_ = nullptr;
}
config_.rtp.local_ssrc = local_ssrc;
config_.rtp.transport_cc = use_transport_cc;
config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
config_.rtp.extensions = extensions;
RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioReceiveStream(config_);
RTC_CHECK(stream_);
SetPlayout(playout_);
}
rtc::ThreadChecker worker_thread_checker_;
webrtc::Call* call_ = nullptr;
webrtc::AudioReceiveStream::Config config_;
// The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
// configuration changes.
webrtc::AudioReceiveStream* stream_ = nullptr;
bool playout_ = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const MediaConfig& config,
const AudioOptions& options,
webrtc::Call* call)
: VoiceMediaChannel(config), engine_(engine), call_(call) {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
engine->RegisterChannel(this);
SetOptions(options);
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
// TODO(solenberg): Should be able to delete the streams directly, without
// going through RemoveNnStream(), once stream objects handle
// all (de)configuration.
while (!send_streams_.empty()) {
RemoveSendStream(send_streams_.begin()->first);
}
while (!recv_streams_.empty()) {
RemoveRecvStream(recv_streams_.begin()->first);
}
engine()->UnregisterChannel(this);
}
rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
return kAudioDscpValue;
}
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
<< params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
if (!SetSendCodecs(params.codecs)) {
return false;
}
if (!ValidateRtpExtensions(params.extensions)) {
return false;
}
std::vector<webrtc::RtpExtension> filtered_extensions =
FilterRtpExtensions(params.extensions,
webrtc::RtpExtension::IsSupportedForAudio, true);
if (send_rtp_extensions_ != filtered_extensions) {
send_rtp_extensions_.swap(filtered_extensions);
for (auto& it : send_streams_) {
it.second->RecreateAudioSendStream(send_rtp_extensions_);
}
}
if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
return false;
}
return SetOptions(params.options);
}
bool WebRtcVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
<< params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
if (!SetRecvCodecs(params.codecs)) {
return false;
}
if (!ValidateRtpExtensions(params.extensions)) {
return false;
}
std::vector<webrtc::RtpExtension> filtered_extensions =
FilterRtpExtensions(params.extensions,
webrtc::RtpExtension::IsSupportedForAudio, false);
if (recv_rtp_extensions_ != filtered_extensions) {
recv_rtp_extensions_.swap(filtered_extensions);
for (auto& it : recv_streams_) {
it.second->RecreateAudioReceiveStream(recv_rtp_extensions_);
}
}
return true;
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
webrtc::RtpParameters rtp_params = it->second->rtp_parameters();
// Need to add the common list of codecs to the send stream-specific
// RTP parameters.
for (const AudioCodec& codec : send_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
// TODO(deadbeef): Handle setting parameters with a list of codecs in a
// different order (which should change the send codec).
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
<< "is not currently supported.";
return false;
}
// TODO(minyue): The following legacy actions go into
// |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
// though there are two difference:
// 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls
// |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls
// |SetSendCodecs|. The outcome should be the same.
// 2. AudioSendStream can be recreated.
// Codecs are handled at the WebRtcVoiceMediaChannel level.
webrtc::RtpParameters reduced_params = parameters;
reduced_params.codecs.clear();
return it->second->SetRtpParameters(reduced_params);
}
webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
// TODO(deadbeef): Return stream-specific parameters.
webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
for (const AudioCodec& codec : recv_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
}
return rtp_params;
}
bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!ValidateRtpParameters(parameters)) {
return false;
}
auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
<< "with ssrc " << ssrc << " which doesn't exist.";
return false;
}
webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
if (current_parameters != parameters) {
LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
<< "unsupported.";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
const webrtc::RtpParameters& rtp_parameters) {
if (rtp_parameters.encodings.size() != 1) {
LOG(LS_ERROR)
<< "Attempted to set RtpParameters without exactly one encoding";
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
// We retain all of the existing options, and apply the given ones
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
options_.SetAll(options);
if (!engine()->ApplyOptions(options_)) {
LOG(LS_WARNING) <<
"Failed to apply engine options during channel SetOptions.";
return false;
}
rtc::Optional<std::string> audio_network_adatptor_config =
GetAudioNetworkAdaptorConfig(options_);
for (auto& it : send_streams_) {
it.second->RecreateAudioSendStream(audio_network_adatptor_config);
}
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
}
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set the payload types to be used for incoming media.
LOG(LS_INFO) << "Setting receive voice codecs.";
if (!VerifyUniquePayloadTypes(codecs)) {
LOG(LS_ERROR) << "Codec payload types overlap.";
return false;
}
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.
ChangePlayout(false);
}
bool result = true;
for (const AudioCodec& codec : new_codecs) {
webrtc::CodecInst voe_codec = {0};
if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
LOG(LS_INFO) << ToString(codec);
voe_codec.pltype = codec.id;
for (const auto& ch : recv_streams_) {
if (engine()->voe()->codec()->SetRecPayloadType(
ch.second->channel(), voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
ToString(voe_codec));
result = false;
}
}
} else {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
result = false;
break;
}
}
if (result) {
recv_codecs_ = codecs;
}
if (desired_playout_ && !playout_) {
ChangePlayout(desired_playout_);
}
return result;
}
// Utility function called from SetSendParameters() to extract current send
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// TODO(solenberg): Validate input - that payload types don't overlap, are
// within range, filter out codecs we don't support,
// redundant codecs etc - the same way it is done for
// RtpHeaderExtensions.
// Find the DTMF telephone event "codec" payload type.
dtmf_payload_type_ = rtc::Optional<int>();
for (const AudioCodec& codec : codecs) {
if (IsCodec(codec, kDtmfCodecName)) {
if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
return false;
}
dtmf_payload_type_ = rtc::Optional<int>(codec.id);
break;
}
}
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD, CNG, NACK and Opus-specific
// parameters.
// TODO(solenberg): Refactor this logic once we create AudioEncoders here.
webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
{
send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
// Find send codec (the first non-telephone-event/CN codec).
const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
codecs, &send_codec_spec.codec_inst);
if (!codec) {
LOG(LS_WARNING) << "Received empty list of codecs.";
return false;
}
send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
send_codec_spec.nack_enabled = HasNack(*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_spec.codec_inst,
&send_codec_spec.enable_codec_fec,
&send_codec_spec.opus_max_playback_rate,
&send_codec_spec.enable_opus_dtx,
&send_codec_spec.min_ptime_ms,
&send_codec_spec.max_ptime_ms);
}
// Set packet size if the AudioCodec param kCodecParamPTime is set.
int ptime_ms = 0;
if (codec->GetParam(kCodecParamPTime, &ptime_ms)) {
if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(
&send_codec_spec.codec_inst, ptime_ms)) {
LOG(LS_WARNING) << "Failed to set packet size for codec "
<< send_codec_spec.codec_inst.plname;
return false;
}
}
// Loop through the codecs list again to find the CN codec.
// TODO(solenberg): Break out into a separate function?
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 = {0};
if (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
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).
int cng_plfreq = -1;
switch (codec.clockrate) {
case 8000:
case 16000:
case 32000:
cng_plfreq = codec.clockrate;
break;
default:
LOG(LS_WARNING) << "CN frequency " << codec.clockrate
<< " not supported.";
continue;
}
send_codec_spec.cng_payload_type = codec.id;
send_codec_spec.cng_plfreq = cng_plfreq;
break;
}
}
}
// Apply new settings to all streams.
if (send_codec_spec_ != send_codec_spec) {
send_codec_spec_ = std::move(send_codec_spec);
for (const auto& kv : send_streams_) {
kv.second->RecreateAudioSendStream(send_codec_spec_);
}
}
// Check if the transport cc feedback or NACK status has changed on the
// preferred send codec, and in that case reconfigure all receive streams.
if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled ||
recv_nack_enabled_ != send_codec_spec_.nack_enabled) {
LOG(LS_INFO) << "Recreate all the receive streams because the send "
"codec has changed.";
recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
recv_nack_enabled_ = send_codec_spec_.nack_enabled;
for (auto& kv : recv_streams_) {
kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_,
recv_nack_enabled_);
}
}
send_codecs_ = codecs;
return true;
}
void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
desired_playout_ = playout;
return ChangePlayout(desired_playout_);
}
void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (playout_ == playout) {
return;
}
for (const auto& kv : recv_streams_) {
kv.second->SetPlayout(playout);
}
playout_ = playout;
}
void WebRtcVoiceMediaChannel::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
if (send_ == send) {
return;
}
// Apply channel specific options, and initialize the ADM for recording (this
// may take time on some platforms, e.g. Android).
if (send) {
engine()->ApplyOptions(options_);
// InitRecording() may return an error if the ADM is already recording.
if (!engine()->adm()->RecordingIsInitialized() &&
!engine()->adm()->Recording()) {
if (engine()->adm()->InitRecording() != 0) {
LOG(LS_WARNING) << "Failed to initialize recording";
}
}
}
// Change the settings on each send channel.
for (auto& kv : send_streams_) {
kv.second->SetSend(send);
}
send_ = send;
}
bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// TODO(solenberg): The state change should be fully rolled back if any one of
// these calls fail.
if (!SetLocalSource(ssrc, source)) {
return false;
}
if (!MuteStream(ssrc, !enable)) {
return false;
}
if (enable && options) {
return SetOptions(*options);
}
return true;
}
int WebRtcVoiceMediaChannel::CreateVoEChannel() {
int id = engine()->CreateVoEChannel();
if (id == -1) {
LOG_RTCERR0(CreateVoEChannel);
return -1;
}
return id;
}
bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
LOG_RTCERR1(DeleteChannel, channel);
return false;
}
return true;
}
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(0 != ssrc);
if (GetSendChannelId(ssrc) != -1) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
// Create a new channel for sending audio data.
int channel = CreateVoEChannel();
if (channel == -1) {
return false;
}
// Save the channel to send_streams_, so that RemoveSendStream() can still
// delete the channel in case failure happens below.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
rtc::Optional<std::string> audio_network_adaptor_config =
GetAudioNetworkAdaptorConfig(options_);
WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
channel, audio_transport, ssrc, sp.cname, send_codec_spec_,
send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config,
call_, this);
send_streams_.insert(std::make_pair(ssrc, stream));
// At this point the stream's local SSRC has been updated. If it is the first
// send stream, make sure that all the receive streams are updated with the
// same SSRC in order to send receiver reports.
if (send_streams_.size() == 1) {
receiver_reports_ssrc_ = ssrc;
for (const auto& kv : recv_streams_) {
// TODO(solenberg): Allow applications to set the RTCP SSRC of receive
// streams instead, so we can avoid recreating the streams here.
kv.second->RecreateAudioReceiveStream(ssrc);
int recv_channel = kv.second->channel();
engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
<< " is associated with channel #" << channel << ".";
}
}
send_streams_[ssrc]->SetSend(send_);
return true;
}
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
it->second->SetSend(false);
// Clean up and delete the send stream+channel.
int channel = it->second->channel();
LOG(LS_INFO) << "Removing audio send stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
delete it->second;
send_streams_.erase(it);
if (!DeleteVoEChannel(channel)) {
return false;
}
if (send_streams_.empty()) {
SetSend(false);
}
return true;
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
if (!ValidateStreamParams(sp)) {
return false;
}
const uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
return false;
}
// Remove the default receive stream if one had been created with this ssrc;
// we'll recreate it then.
if (IsDefaultRecvStream(ssrc)) {
RemoveRecvStream(ssrc);
}
if (GetReceiveChannelId(ssrc) != -1) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
// Create a new channel for receiving audio data.
const int channel = CreateVoEChannel();
if (channel == -1) {
return false;
}
// Turn off all supported codecs.
// TODO(solenberg): Remove once "no codecs" is the default state of a stream.
for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
voe_codec.pltype = -1;
if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
DeleteVoEChannel(channel);
return false;
}
}
// Only enable those configured for this channel.
for (const auto& codec : recv_codecs_) {
webrtc::CodecInst voe_codec = {0};
if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
voe_codec.pltype = codec.id;
if (engine()->voe()->codec()->SetRecPayloadType(
channel, voe_codec) == -1) {
LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
DeleteVoEChannel(channel);
return false;
}
}
}
const int send_channel = GetSendChannelId(receiver_reports_ssrc_);
if (send_channel != -1) {
// Associate receive channel with first send channel (so the receive channel
// can obtain RTT from the send channel)
engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
LOG(LS_INFO) << "VoiceEngine channel #" << channel
<< " is associated with channel #" << send_channel << ".";
}
recv_streams_.insert(std::make_pair(
ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
recv_transport_cc_enabled_,
recv_nack_enabled_,
sp.sync_label, recv_rtp_extensions_,
call_, this,
engine()->decoder_factory_)));
recv_streams_[ssrc]->SetPlayout(playout_);
return true;
}
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
// Deregister default channel, if that's the one being destroyed.
if (IsDefaultRecvStream(ssrc)) {
default_recv_ssrc_ = -1;
}
const int channel = it->second->channel();
// Clean up and delete the receive stream+channel.
LOG(LS_INFO) << "Removing audio receive stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
it->second->SetRawAudioSink(nullptr);
delete it->second;
recv_streams_.erase(it);
return DeleteVoEChannel(channel);
}
bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
AudioSource* source) {
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
if (source) {
// Return an error if trying to set a valid source with an invalid ssrc.
LOG(LS_ERROR) << "SetLocalSource failed with ssrc " << ssrc;
return false;
}
// The channel likely has gone away, do nothing.
return true;
}
if (source) {
it->second->SetSource(source);
} else {
it->second->ClearSource();
}
return true;
}
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
actives->clear();
for (const auto& ch : recv_streams_) {
int level = GetOutputLevel(ch.second->channel());
if (level > 0) {
actives->push_back(std::make_pair(ch.first, level));
}
}
return true;
}
int WebRtcVoiceMediaChannel::GetOutputLevel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int highest = 0;
for (const auto& ch : recv_streams_) {
highest = std::max(GetOutputLevel(ch.second->channel()), 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::SetOutputVolume(uint32_t ssrc, double volume) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (ssrc == 0) {
default_recv_volume_ = volume;
if (default_recv_ssrc_ == -1) {
return true;
}
ssrc = static_cast<uint32_t>(default_recv_ssrc_);
}
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "SetOutputVolume: no recv stream" << ssrc;
return false;
}
it->second->SetOutputVolume(volume);
LOG(LS_INFO) << "SetOutputVolume() to " << volume
<< " for recv stream with ssrc " << ssrc;
return true;
}
bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
return dtmf_payload_type_ ? true : false;
}
bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
int duration) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
if (!dtmf_payload_type_) {
return false;
}
// Figure out which WebRtcAudioSendStream to send the event on.
auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin();
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
if (event < kMinTelephoneEventCode ||
event > kMaxTelephoneEventCode) {
LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
return false;
}
if (duration < kMinTelephoneEventDuration ||
duration > kMaxTelephoneEventDuration) {
LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
return false;
}
return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration);
}
void WebRtcVoiceMediaChannel::OnPacketReceived(
rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
webrtc::PacketReceiver::DeliveryStatus delivery_result =
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
packet->cdata(), packet->size(),
webrtc_packet_time);
if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
return;
}
// Create a default receive stream for this unsignalled and previously not
// received ssrc. If there already is a default receive stream, delete it.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
if (default_recv_ssrc_ != -1) {
LOG(LS_INFO) << "Removing default receive stream with ssrc "
<< default_recv_ssrc_;
RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
RemoveRecvStream(default_recv_ssrc_);
default_recv_ssrc_ = -1;
}
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
if (!AddRecvStream(sp)) {
LOG(LS_WARNING) << "Could not create default receive stream.";
return;
}
default_recv_ssrc_ = ssrc;
SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
if (default_sink_) {
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
new ProxySink(default_sink_.get()));
SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
}
delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
packet->cdata(),
packet->size(),
webrtc_packet_time);
RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Forward packet to Call as well.
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
packet->cdata(), packet->size(), webrtc_packet_time);
}
void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
call_->OnNetworkRouteChanged(transport_name, network_route);
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
it->second->SetMuted(muted);
// TODO(solenberg):
// We set the AGC to mute state only when all the channels are muted.
// This implementation is not ideal, instead we should signal the AGC when
// the mic channel is muted/unmuted. We can't do it today because there
// is no good way to know which stream is mapping to the mic channel.
bool all_muted = muted;
for (const auto& kv : send_streams_) {
all_muted = all_muted && kv.second->muted();
}
engine()->apm()->set_output_will_be_muted(all_muted);
return true;
}
bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate.";
max_send_bitrate_bps_ = bps;
bool success = true;
for (const auto& kv : send_streams_) {
if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) {
success = false;
}
}
return success;
}
void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
call_->SignalChannelNetworkState(
webrtc::MediaType::AUDIO,
ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}
void WebRtcVoiceMediaChannel::OnTransportOverheadChanged(
int transport_overhead_per_packet) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
transport_overhead_per_packet);
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(info);
// Get SSRC and stats for each sender.
RTC_DCHECK(info->senders.size() == 0);
for (const auto& stream : send_streams_) {
webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
VoiceSenderInfo sinfo;
sinfo.add_ssrc(stats.local_ssrc);
sinfo.bytes_sent = stats.bytes_sent;
sinfo.packets_sent = stats.packets_sent;
sinfo.packets_lost = stats.packets_lost;
sinfo.fraction_lost = stats.fraction_lost;
sinfo.codec_name = stats.codec_name;
sinfo.ext_seqnum = stats.ext_seqnum;
sinfo.jitter_ms = stats.jitter_ms;
sinfo.rtt_ms = stats.rtt_ms;
sinfo.audio_level = stats.audio_level;
sinfo.aec_quality_min = stats.aec_quality_min;
sinfo.echo_delay_median_ms = stats.echo_delay_median_ms;
sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
sinfo.echo_return_loss = stats.echo_return_loss;
sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
sinfo.residual_echo_likelihood = stats.residual_echo_likelihood;
sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
info->senders.push_back(sinfo);
}
// Get SSRC and stats for each receiver.
RTC_DCHECK(info->receivers.size() == 0);
for (const auto& stream : recv_streams_) {
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
VoiceReceiverInfo rinfo;
rinfo.add_ssrc(stats.remote_ssrc);
rinfo.bytes_rcvd = stats.bytes_rcvd;
rinfo.packets_rcvd = stats.packets_rcvd;
rinfo.packets_lost = stats.packets_lost;
rinfo.fraction_lost = stats.fraction_lost;
rinfo.codec_name = stats.codec_name;
rinfo.ext_seqnum = stats.ext_seqnum;
rinfo.jitter_ms = stats.jitter_ms;
rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
rinfo.delay_estimate_ms = stats.delay_estimate_ms;
rinfo.audio_level = stats.audio_level;
rinfo.expand_rate = stats.expand_rate;
rinfo.speech_expand_rate = stats.speech_expand_rate;
rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
rinfo.accelerate_rate = stats.accelerate_rate;
rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
rinfo.decoding_calls_to_silence_generator =
stats.decoding_calls_to_silence_generator;
rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
rinfo.decoding_normal = stats.decoding_normal;
rinfo.decoding_plc = stats.decoding_plc;
rinfo.decoding_cng = stats.decoding_cng;
rinfo.decoding_plc_cng = stats.decoding_plc_cng;
rinfo.decoding_muted_output = stats.decoding_muted_output;
rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
info->receivers.push_back(rinfo);
}
return true;
}
void WebRtcVoiceMediaChannel::SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
<< " " << (sink ? "(ptr)" : "NULL");
if (ssrc == 0) {
if (default_recv_ssrc_ != -1) {
std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
sink ? new ProxySink(sink.get()) : nullptr);
SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
}
default_sink_ = std::move(sink);
return;
}
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
return;
}
it->second->SetRawAudioSink(std::move(sink));
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
unsigned int ulevel = 0;
int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
return (ret == 0) ? static_cast<int>(ulevel) : -1;
}
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = recv_streams_.find(ssrc);
if (it != recv_streams_.end()) {
return it->second->channel();
}
return -1;
}
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = send_streams_.find(ssrc);
if (it != send_streams_.end()) {
return it->second->channel();
}
return -1;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE