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webrtc_m130/webrtc/media/engine/webrtcvoiceengine.cc
skvlad 7a43d253f9 Make the audio channel communicate network state changes to the call. 
This change enables voice-only calls to keep track of the network state.
This is only a partial fix - the last modality to change state controls
the state for the entire call, so a call with a failed video transport
will also stop sending audio packets. Handling this condition correctly
would require the call to keep track of network state for each media
type separately, and take care of conditions such as a failed video
channel getting removed, while a functioning audio channel remains.

BUG=webrtc:5307

Review URL: https://codereview.webrtc.org/1757683002

Cr-Commit-Position: refs/heads/master@{#12093}
2016-03-22 22:32:31 +00:00

2558 lines
89 KiB
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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_CONFIG_H
#include <config.h>
#endif
#ifdef HAVE_WEBRTC_VOICE
#include "webrtc/media/engine/webrtcvoiceengine.h"
#include <algorithm>
#include <cstdio>
#include <string>
#include <vector>
#include "webrtc/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/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/media/base/audioframe.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.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
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
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.
const int kOpusBitrateNb = 12000;
const int kOpusBitrateWb = 20000;
const int kOpusBitrateFb = 32000;
// Opus bitrate should be in the range between 6000 and 510000.
const int kOpusMinBitrate = 6000;
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
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;
}
// 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 = 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.
int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
int value;
if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
return value;
}
return kOpusDefaultMaxPlaybackRate;
}
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);
}
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> SupportedCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
std::vector<AudioCodec> result;
for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
// Change the sample rate of G722 to 8000 to match SDP.
MaybeFixupG722(&voe_codec, 8000);
// Skip uncompressed formats.
if (IsCodec(voe_codec, kL16CodecName)) {
continue;
}
const CodecPref* pref = NULL;
for (size_t j = 0; j < arraysize(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,
static_cast<int>(arraysize(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);
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);
} else {
LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
}
}
// Make sure they are in local preference order.
std::sort(result.begin(), result.end(), &AudioCodec::Preferable);
return result;
}
static bool ToCodecInst(const AudioCodec& in,
webrtc::CodecInst* out) {
for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
// Change the sample rate of G722 to 8000 to match SDP.
MaybeFixupG722(&voe_codec, 8000);
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
bool multi_rate = IsCodecMultiRate(voe_codec);
// Allow arbitrary rates for ISAC to be specified.
if (multi_rate) {
// Set codec.bitrate to 0 so the check for codec.Matches() passes.
codec.bitrate = 0;
}
if (codec.Matches(in)) {
if (out) {
// Fixup the payload type.
voe_codec.pltype = in.id;
// Set bitrate if specified.
if (multi_rate && in.bitrate != 0) {
voe_codec.rate = in.bitrate;
}
// Reset G722 sample rate to 16000 to match WebRTC.
MaybeFixupG722(&voe_codec, 16000);
// Apply codec-specific settings.
if (IsCodec(codec, kIsacCodecName)) {
// If ISAC and an explicit bitrate is not specified,
// enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
}
return true;
}
}
return false;
}
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;
}
// 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,
int* red_payload_type) {
RTC_DCHECK(out);
RTC_DCHECK(red_payload_type);
// Select the preferred send codec (the first non-telephone-event/CN codec).
for (const AudioCodec& codec : codecs) {
*red_payload_type = -1;
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.
const AudioCodec* found_codec = &codec;
if (IsCodec(*found_codec, kRedCodecName)) {
// Parse out the RED parameters. If we fail, just ignore RED;
// we don't support all possible params/usage scenarios.
*red_payload_type = codec.id;
found_codec = GetRedSendCodec(*found_codec, codecs);
if (!found_codec) {
continue;
}
}
// 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 (!ToCodecInst(*found_codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(*found_codec);
continue;
}
*out = voe_codec;
return found_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];
};
// Note: keep the supported packet sizes in ascending order.
static const CodecPref kCodecPrefs[12];
static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
for (int packet_size_ms : codec_pref.packet_sizes_ms) {
if (packet_size_ms && packet_size_ms <= ptime_ms) {
selected_packet_size_ms = packet_size_ms;
}
}
return selected_packet_size_ms;
}
// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
// codec.
static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
if (IsCodec(*voe_codec, kG722CodecName)) {
// If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
// has changed, and this special case is no longer needed.
RTC_DCHECK(voe_codec->plfreq != new_plfreq);
voe_codec->plfreq = new_plfreq;
}
}
static const AudioCodec* GetRedSendCodec(
const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs) {
// 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 nullptr;
}
} 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) {
return &codec;
}
}
LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
return nullptr;
}
};
const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
{ kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
{ kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
{ kIsacCodecName, 32000, 1, 104, true, { 30 } },
// G722 should be advertised as 8000 Hz because of the RFC "bug".
{ kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
{ kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
{ kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
{ kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
{ kCnCodecName, 32000, 1, 106, false, { } },
{ kCnCodecName, 16000, 1, 105, false, { } },
{ kCnCodecName, 8000, 1, 13, false, { } },
{ kRedCodecName, 8000, 1, 127, false, { } },
{ kDtmfCodecName, 8000, 1, 126, false, { } },
};
} // namespace {
bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
webrtc::CodecInst* out) {
return WebRtcVoiceCodecs::ToCodecInst(in, out);
}
WebRtcVoiceEngine::WebRtcVoiceEngine()
: voe_wrapper_(new VoEWrapper()),
audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))) {
Construct();
}
WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper)
: voe_wrapper_(voe_wrapper) {
Construct();
}
void WebRtcVoiceEngine::Construct() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
signal_thread_checker_.DetachFromThread();
std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
webrtc::Trace::SetTraceCallback(this);
// Load our audio codec list.
codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
if (adm_) {
voe_wrapper_.reset();
adm_->Release();
adm_ = NULL;
}
webrtc::Trace::SetTraceCallback(nullptr);
}
bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(worker_thread == rtc::Thread::Current());
LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
bool res = InitInternal();
if (res) {
LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
} else {
LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
Terminate();
}
return res;
}
bool WebRtcVoiceEngine::InitInternal() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Temporarily turn logging level up for the Init call.
webrtc::Trace::set_level_filter(kElevatedTraceFilter);
LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
if (voe_wrapper_->base()->Init(adm_) == -1) {
LOG_RTCERR0_EX(Init, voe_wrapper_->error());
return false;
}
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
// Save the default AGC configuration settings. This must happen before
// calling ApplyOptions or the default will be overwritten.
if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
LOG_RTCERR0(GetAgcConfig);
return false;
}
// 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);
if (!ApplyOptions(options)) {
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);
}
SetDefaultDevices();
initialized_ = true;
return true;
}
void WebRtcVoiceEngine::Terminate() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
initialized_ = false;
StopAecDump();
voe_wrapper_->base()->Terminate();
}
rtc::scoped_refptr<webrtc::AudioState>
WebRtcVoiceEngine::GetAudioState() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return audio_state_;
}
VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
webrtc::Call* call,
const 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) << "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 and AGC.
options.echo_cancellation = rtc::Optional<bool>(false);
options.auto_gain_control = rtc::Optional<bool>(false);
LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
#elif defined(ANDROID)
ec_mode = webrtc::kEcAecm;
#endif
#if defined(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
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 = 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 =
*options.echo_cancellation && !use_delay_agnostic_aec;
if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
options.echo_cancellation = 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) {
const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
if (built_in_agc) {
if (voe_wrapper_->hw()->EnableBuiltInAGC(*options.auto_gain_control) ==
0 &&
*options.auto_gain_control) {
// Disable internal software AGC if built-in AGC is enabled,
// i.e., replace the software AGC with the built-in AGC.
options.auto_gain_control = 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.noise_suppression) {
const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
if (built_in_ns) {
if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) ==
0 &&
*options.noise_suppression) {
// Disable internal software NS if built-in NS is enabled,
// i.e., replace the software NS with the built-in NS.
options.noise_suppression = 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;
voe_config_.Set<webrtc::NetEqCapacityConfig>(
new webrtc::NetEqCapacityConfig(
*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;
voe_config_.Set<webrtc::NetEqFastAccelerate>(
new webrtc::NetEqFastAccelerate(
*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_));
}
// 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);
}
if (options.recording_sample_rate) {
LOG(LS_INFO) << "Recording sample rate is "
<< *options.recording_sample_rate;
if (voe_wrapper_->hw()->SetRecordingSampleRate(
*options.recording_sample_rate)) {
LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
}
}
if (options.playout_sample_rate) {
LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
if (voe_wrapper_->hw()->SetPlayoutSampleRate(
*options.playout_sample_rate)) {
LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
}
}
return true;
}
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;
}
webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
if (ap) {
ap->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
}
bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
unsigned int ulevel;
if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
LOG_RTCERR1(GetSpeakerVolume, level);
return false;
}
*level = ulevel;
return true;
}
bool WebRtcVoiceEngine::SetOutputVolume(int level) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(level >= 0 && level <= 255);
if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
LOG_RTCERR1(SetSpeakerVolume, level);
return false;
}
return true;
}
int WebRtcVoiceEngine::GetInputLevel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
unsigned int ulevel;
return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
static_cast<int>(ulevel) : -1;
}
const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
return codecs_;
}
RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
RtpCapabilities capabilities;
capabilities.header_extensions.push_back(RtpHeaderExtension(
kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
capabilities.header_extensions.push_back(
RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
"Enabled") {
capabilities.header_extensions.push_back(RtpHeaderExtension(
kRtpTransportSequenceNumberHeaderExtension,
kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
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::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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,
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 (voe_wrapper_->base()->audio_processing()->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 (voe_wrapper_->base()->audio_processing()->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 (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
webrtc::AudioProcessing::kNoError) {
LOG_RTCERR0(StopDebugRecording);
}
is_dumping_aec_ = false;
}
}
bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
if (event_log) {
return event_log->StartLogging(file);
}
LOG_RTCERR0(StartRtcEventLog);
return false;
}
void WebRtcVoiceEngine::StopRtcEventLog() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
if (event_log) {
event_log->StopLogging();
return;
}
LOG_RTCERR0(StopRtcEventLog);
}
int WebRtcVoiceEngine::CreateVoEChannel() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->base()->CreateChannel(voe_config_);
}
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioSource::Sink {
public:
WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
uint32_t ssrc, const std::string& c_name,
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call)
: voe_audio_transport_(voe_audio_transport),
call_(call),
config_(nullptr) {
RTC_DCHECK_GE(ch, 0);
// TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
// RTC_DCHECK(voe_audio_transport);
RTC_DCHECK(call);
audio_capture_thread_checker_.DetachFromThread();
config_.rtp.ssrc = ssrc;
config_.rtp.c_name = c_name;
config_.voe_channel_id = ch;
RecreateAudioSendStream(extensions);
}
~WebRtcAudioSendStream() override {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
ClearSource();
call_->DestroyAudioSendStream(stream_);
}
void RecreateAudioSendStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioSendStream(stream_);
stream_ = nullptr;
}
config_.rtp.extensions = extensions;
RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioSendStream(config_);
RTC_CHECK(stream_);
}
bool SendTelephoneEvent(int payload_type, 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();
}
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_DCHECK(!worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
RTC_DCHECK(voe_audio_transport_);
voe_audio_transport_->OnData(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;
}
private:
void UpdateSendState() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
if (send_ && source_ != nullptr) {
stream_->Start();
} else { // !send || source_ = nullptr
stream_->Stop();
}
}
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker audio_capture_thread_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;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};
class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
public:
WebRtcAudioReceiveStream(int ch,
uint32_t remote_ssrc,
uint32_t local_ssrc,
bool use_transport_cc,
const std::string& sync_group,
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call)
: call_(call), config_() {
RTC_DCHECK_GE(ch, 0);
RTC_DCHECK(call);
config_.rtp.remote_ssrc = remote_ssrc;
config_.rtp.local_ssrc = local_ssrc;
config_.voe_channel_id = ch;
config_.sync_group = sync_group;
RecreateAudioReceiveStream(use_transport_cc, extensions);
}
~WebRtcAudioReceiveStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
call_->DestroyAudioReceiveStream(stream_);
}
void RecreateAudioReceiveStream(
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RecreateAudioReceiveStream(config_.rtp.transport_cc, extensions);
}
void RecreateAudioReceiveStream(bool use_transport_cc) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RecreateAudioReceiveStream(use_transport_cc, config_.rtp.extensions);
}
webrtc::AudioReceiveStream::Stats GetStats() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(stream_);
return stream_->GetStats();
}
int channel() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_.voe_channel_id;
}
void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
stream_->SetSink(std::move(sink));
}
private:
void RecreateAudioReceiveStream(
bool use_transport_cc,
const std::vector<webrtc::RtpExtension>& extensions) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (stream_) {
call_->DestroyAudioReceiveStream(stream_);
stream_ = nullptr;
}
config_.rtp.extensions = extensions;
config_.rtp.transport_cc = use_transport_cc;
RTC_DCHECK(!stream_);
stream_ = call_->CreateAudioReceiveStream(config_);
RTC_CHECK(stream_);
}
rtc::ThreadChecker worker_thread_checker_;
webrtc::Call* call_ = nullptr;
webrtc::AudioReceiveStream::Config config_;
// The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
// configuration changes.
webrtc::AudioReceiveStream* stream_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(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 (!SetMaxSendBandwidth(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;
}
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;
}
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.
PausePlayout();
}
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_) {
ResumePlayout();
}
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, RED, NACK and Opus-specific
// parameters.
{
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, &send_codec_spec.red_payload_type);
if (!codec) {
LOG(LS_WARNING) << "Received empty list of codecs.";
return false;
}
send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
// This condition is apparently here because Opus does not support RED and
// FEC simultaneously. However, DTX and max playback rate shouldn't have
// such limitations.
// TODO(solenberg): Refactor this logic once we create AudioEncoders here.
if (send_codec_spec.red_payload_type == -1) {
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);
}
// 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;
}
}
// Latch in the new state.
send_codec_spec_ = std::move(send_codec_spec);
}
// Cache the codecs in order to configure the channel created later.
for (const auto& ch : send_streams_) {
if (!SetSendCodecs(ch.second->channel())) {
return false;
}
}
// Set nack status on receive channels.
if (!send_streams_.empty()) {
for (const auto& kv : recv_streams_) {
SetNack(kv.second->channel(), send_codec_spec_.nack_enabled);
}
}
// Check if the transport cc feedback 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) {
LOG(LS_INFO) << "Recreate all the receive streams because the send "
"codec has changed.";
recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
for (auto& kv : recv_streams_) {
kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_);
}
}
return true;
}
// Apply current codec settings to a single voe::Channel used for sending.
bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
// 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);
if (send_codec_spec_.red_payload_type != -1) {
// 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,
send_codec_spec_.red_payload_type) == -1) {
LOG_RTCERR3(SetREDStatus, channel, true,
send_codec_spec_.red_payload_type);
return false;
}
}
SetNack(channel, send_codec_spec_.nack_enabled);
// Set the codec immediately, since SetVADStatus() depends on whether
// the current codec is mono or stereo.
if (!SetSendCodec(channel, send_codec_spec_.codec_inst)) {
return false;
}
// FEC should be enabled after SetSendCodec.
if (send_codec_spec_.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_spec_.codec_inst, 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 "
<< (send_codec_spec_.enable_opus_dtx ? "enable" : "disable")
<< " Opus DTX on channel "
<< channel;
if (engine()->voe()->codec()->SetOpusDtx(channel,
send_codec_spec_.enable_opus_dtx)) {
LOG_RTCERR2(SetOpusDtx, channel, send_codec_spec_.enable_opus_dtx);
return false;
}
// If opus_max_playback_rate <= 0, the default maximum playback rate
// (48 kHz) will be used.
if (send_codec_spec_.opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
<< send_codec_spec_.opus_max_playback_rate
<< " Hz on channel "
<< channel;
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
channel, send_codec_spec_.opus_max_playback_rate) == -1) {
LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
send_codec_spec_.opus_max_playback_rate);
return false;
}
}
}
if (send_bitrate_setting_) {
SetSendBitrateInternal(send_bitrate_bps_);
}
// Set the CN payloadtype and the VAD status.
if (send_codec_spec_.cng_payload_type != -1) {
// The CN payload type for 8000 Hz clockrate is fixed at 13.
if (send_codec_spec_.cng_plfreq != 8000) {
webrtc::PayloadFrequencies cn_freq;
switch (send_codec_spec_.cng_plfreq) {
case 16000:
cn_freq = webrtc::kFreq16000Hz;
break;
case 32000:
cn_freq = webrtc::kFreq32000Hz;
break;
default:
RTC_NOTREACHED();
return false;
}
if (engine()->voe()->codec()->SetSendCNPayloadType(
channel, send_codec_spec_.cng_payload_type, cn_freq) == -1) {
LOG_RTCERR3(SetSendCNPayloadType, channel,
send_codec_spec_.cng_payload_type, 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 (send_codec_spec_.cng_plfreq == send_codec_spec_.codec_inst.plfreq &&
send_codec_spec_.codec_inst.channels == 1) {
// 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;
}
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(
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 = {0};
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::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) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (playout_ == playout) {
return true;
}
for (const auto& ch : recv_streams_) {
if (!SetPlayout(ch.second->channel(), playout)) {
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
<< ch.second->channel() << " failed";
return false;
}
}
playout_ = playout;
return true;
}
void WebRtcVoiceMediaChannel::SetSend(bool send) {
TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
if (send_ == send) {
return;
}
// Apply channel specific options when channel is enabled for sending.
if (send) {
engine()->ApplyOptions(options_);
}
// 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;
}
if (engine()->voe()->network()->RegisterExternalTransport(id, *this) == -1) {
LOG_RTCERR2(RegisterExternalTransport, id, this);
engine()->voe()->base()->DeleteChannel(id);
return -1;
}
return id;
}
bool WebRtcVoiceMediaChannel::DeleteVoEChannel(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) {
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();
send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream(
channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_)));
// Set the current codecs to be used for the new channel. We need to do this
// after adding the channel to send_channels_, because of how max bitrate is
// currently being configured by SetSendCodec().
if (HasSendCodec() && !SetSendCodecs(channel)) {
RemoveSendStream(ssrc);
return false;
}
// At this point the channel's local SSRC has been updated. If the channel is
// the first send channel make sure that all the receive channels are updated
// with the same SSRC in order to send receiver reports.
if (send_streams_.size() == 1) {
receiver_reports_ssrc_ = ssrc;
for (const auto& stream : recv_streams_) {
int recv_channel = stream.second->channel();
if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) {
LOG_RTCERR2(SetLocalSSRC, recv_channel, ssrc);
return false;
}
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_,
sp.sync_label, recv_rtp_extensions_,
call_)));
SetNack(channel, send_codec_spec_.nack_enabled);
SetPlayout(channel, 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_);
}
int ch_id = GetReceiveChannelId(ssrc);
if (ch_id < 0) {
LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
return false;
}
if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(ch_id,
volume)) {
LOG_RTCERR2(SetChannelOutputVolumeScaling, ch_id, volume);
return false;
}
LOG(LS_INFO) << "SetOutputVolume to " << volume
<< " for channel " << ch_id << " and 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());
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
// If we don't have a default channel, and the SSRC is unknown, create a
// default channel.
if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
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));
}
}
// Forward packet to Call. If the SSRC is unknown we'll return after this.
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 (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
// If the SSRC is unknown here, route it to the default channel, if we have
// one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
if (default_recv_ssrc_ == -1) {
return;
} else {
ssrc = default_recv_ssrc_;
}
}
// Find the channel to send this packet to. It must exist since webrtc::Call
// was able to demux the packet.
int channel = GetReceiveChannelId(ssrc);
RTC_DCHECK(channel != -1);
// Pass it off to the decoder.
engine()->voe()->network()->ReceivedRTPPacket(
channel, packet->cdata(), packet->size(), webrtc_packet_time);
}
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);
// 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->cdata(), packet->size(), &type)) {
LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
return;
}
// If it is a sender report, find the receive channel that is listening.
if (type == kRtcpTypeSR) {
uint32_t ssrc = 0;
if (!GetRtcpSsrc(packet->cdata(), packet->size(), &ssrc)) {
return;
}
int recv_channel_id = GetReceiveChannelId(ssrc);
if (recv_channel_id != -1) {
engine()->voe()->network()->ReceivedRTCPPacket(
recv_channel_id, packet->cdata(), packet->size());
}
}
// 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_streams_) {
engine()->voe()->network()->ReceivedRTCPPacket(
ch.second->channel(), packet->cdata(), packet->size());
}
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int channel = GetSendChannelId(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_streams_) {
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 (!HasSendCodec()) {
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_spec_.codec_inst;
bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
if (is_multi_rate) {
// If codec is multi-rate then just set the bitrate.
codec.rate = bps;
for (const auto& ch : send_streams_) {
if (!SetSendCodec(ch.second->channel(), 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;
}
}
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);
}
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.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.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;
}
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;
}
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE
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