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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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1073 } | 1073 } |
1074 | 1074 |
1075 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream | 1075 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
1076 : public AudioSource::Sink { | 1076 : public AudioSource::Sink { |
1077 public: | 1077 public: |
1078 WebRtcAudioSendStream(int ch, | 1078 WebRtcAudioSendStream(int ch, |
1079 webrtc::AudioTransport* voe_audio_transport, | 1079 webrtc::AudioTransport* voe_audio_transport, |
1080 uint32_t ssrc, | 1080 uint32_t ssrc, |
1081 const std::string& c_name, | 1081 const std::string& c_name, |
1082 const std::vector<webrtc::RtpExtension>& extensions, | 1082 const std::vector<webrtc::RtpExtension>& extensions, |
1083 webrtc::Call* call) | 1083 webrtc::Call* call, |
1084 webrtc::Transport* send_transport) | |
1084 : voe_audio_transport_(voe_audio_transport), | 1085 : voe_audio_transport_(voe_audio_transport), |
1085 call_(call), | 1086 call_(call), |
1086 config_(nullptr), | 1087 config_(send_transport), |
1087 rtp_parameters_(CreateRtpParametersWithOneEncoding()) { | 1088 rtp_parameters_(CreateRtpParametersWithOneEncoding()) { |
1088 RTC_DCHECK_GE(ch, 0); | 1089 RTC_DCHECK_GE(ch, 0); |
1089 // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore: | 1090 // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore: |
1090 // RTC_DCHECK(voe_audio_transport); | 1091 // RTC_DCHECK(voe_audio_transport); |
1091 RTC_DCHECK(call); | 1092 RTC_DCHECK(call); |
1092 audio_capture_thread_checker_.DetachFromThread(); | 1093 audio_capture_thread_checker_.DetachFromThread(); |
1093 config_.rtp.ssrc = ssrc; | 1094 config_.rtp.ssrc = ssrc; |
1094 config_.rtp.c_name = c_name; | 1095 config_.rtp.c_name = c_name; |
1095 config_.voe_channel_id = ch; | 1096 config_.voe_channel_id = ch; |
1096 RecreateAudioSendStream(extensions); | 1097 RecreateAudioSendStream(extensions); |
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1904 } | 1905 } |
1905 return true; | 1906 return true; |
1906 } | 1907 } |
1907 | 1908 |
1908 int WebRtcVoiceMediaChannel::CreateVoEChannel() { | 1909 int WebRtcVoiceMediaChannel::CreateVoEChannel() { |
1909 int id = engine()->CreateVoEChannel(); | 1910 int id = engine()->CreateVoEChannel(); |
1910 if (id == -1) { | 1911 if (id == -1) { |
1911 LOG_RTCERR0(CreateVoEChannel); | 1912 LOG_RTCERR0(CreateVoEChannel); |
1912 return -1; | 1913 return -1; |
1913 } | 1914 } |
1914 if (engine()->voe()->network()->RegisterExternalTransport(id, *this) == -1) { | 1915 |
1915 LOG_RTCERR2(RegisterExternalTransport, id, this); | |
1916 engine()->voe()->base()->DeleteChannel(id); | |
1917 return -1; | |
1918 } | |
1919 return id; | 1916 return id; |
1920 } | 1917 } |
1921 | 1918 |
1922 bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) { | 1919 bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) { |
1923 if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) { | |
1924 LOG_RTCERR1(DeRegisterExternalTransport, channel); | |
1925 } | |
1926 if (engine()->voe()->base()->DeleteChannel(channel) == -1) { | 1920 if (engine()->voe()->base()->DeleteChannel(channel) == -1) { |
1927 LOG_RTCERR1(DeleteChannel, channel); | 1921 LOG_RTCERR1(DeleteChannel, channel); |
1928 return false; | 1922 return false; |
1929 } | 1923 } |
1930 return true; | 1924 return true; |
1931 } | 1925 } |
1932 | 1926 |
1933 bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { | 1927 bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { |
1934 TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream"); | 1928 TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream"); |
1935 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1929 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
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1946 // Create a new channel for sending audio data. | 1940 // Create a new channel for sending audio data. |
1947 int channel = CreateVoEChannel(); | 1941 int channel = CreateVoEChannel(); |
1948 if (channel == -1) { | 1942 if (channel == -1) { |
1949 return false; | 1943 return false; |
1950 } | 1944 } |
1951 | 1945 |
1952 // Save the channel to send_streams_, so that RemoveSendStream() can still | 1946 // Save the channel to send_streams_, so that RemoveSendStream() can still |
1953 // delete the channel in case failure happens below. | 1947 // delete the channel in case failure happens below. |
1954 webrtc::AudioTransport* audio_transport = | 1948 webrtc::AudioTransport* audio_transport = |
1955 engine()->voe()->base()->audio_transport(); | 1949 engine()->voe()->base()->audio_transport(); |
1950 | |
1956 WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( | 1951 WebRtcAudioSendStream* stream = new WebRtcAudioSendStream( |
1957 channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_); | 1952 channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_, |
1953 this); | |
1958 send_streams_.insert(std::make_pair(ssrc, stream)); | 1954 send_streams_.insert(std::make_pair(ssrc, stream)); |
1959 | 1955 |
1960 // Set the current codecs to be used for the new channel. We need to do this | 1956 // Set the current codecs to be used for the new channel. We need to do this |
1961 // after adding the channel to send_channels_, because of how max bitrate is | 1957 // after adding the channel to send_channels_, because of how max bitrate is |
1962 // currently being configured by SetSendCodec(). | 1958 // currently being configured by SetSendCodec(). |
1963 if (HasSendCodec() && !SetSendCodecs(channel, stream->rtp_parameters())) { | 1959 if (HasSendCodec() && !SetSendCodecs(channel, stream->rtp_parameters())) { |
1964 RemoveSendStream(ssrc); | 1960 RemoveSendStream(ssrc); |
1965 return false; | 1961 return false; |
1966 } | 1962 } |
1967 | 1963 |
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2244 LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range."; | 2240 LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range."; |
2245 return false; | 2241 return false; |
2246 } | 2242 } |
2247 return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration); | 2243 return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration); |
2248 } | 2244 } |
2249 | 2245 |
2250 void WebRtcVoiceMediaChannel::OnPacketReceived( | 2246 void WebRtcVoiceMediaChannel::OnPacketReceived( |
2251 rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { | 2247 rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { |
2252 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2248 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
2253 | 2249 |
2250 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, | |
2251 packet_time.not_before); | |
2252 webrtc::PacketReceiver::DeliveryStatus delivery_result = | |
2253 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, | |
2254 packet->cdata(), packet->size(), | |
2255 webrtc_packet_time); | |
2256 | |
the sun
2016/04/28 09:15:56
nit: remove blank line
mflodman
2016/04/29 05:54:00
Done.
| |
2257 if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) { | |
2258 return; | |
2259 } | |
2260 | |
2261 // Create a default receive stream for this unsignalled annd previously not | |
the sun
2016/04/28 09:15:56
s/annd/and
mflodman
2016/04/29 05:54:00
Done.
| |
2262 // received ssrc. If there already is a default receive stream, delete it. | |
the sun
2016/04/28 09:15:56
Add:
// See: https://bugs.chromium.org/p/webrtc/is
mflodman
2016/04/29 05:54:00
Done.
| |
2254 uint32_t ssrc = 0; | 2263 uint32_t ssrc = 0; |
2255 if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { | 2264 if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { |
2256 return; | 2265 return; |
2257 } | 2266 } |
2258 | 2267 |
2259 // If we don't have a default channel, and the SSRC is unknown, create a | 2268 if (default_recv_ssrc_ != -1) { |
the sun
2016/04/28 09:15:56
Thanks! This logic is actually easier to follow no
mflodman
2016/04/29 05:54:00
Agree, thanks!
| |
2260 // default channel. | 2269 LOG(LS_INFO) << "Removing default receive stream with ssrc " |
2261 if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) { | 2270 << default_recv_ssrc_; |
2262 StreamParams sp; | 2271 RTC_DCHECK_NE(ssrc, default_recv_ssrc_); |
2263 sp.ssrcs.push_back(ssrc); | 2272 RemoveRecvStream(default_recv_ssrc_); |
2264 LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; | 2273 default_recv_ssrc_ = -1; |
2265 if (!AddRecvStream(sp)) { | |
2266 LOG(LS_WARNING) << "Could not create default receive stream."; | |
2267 return; | |
2268 } | |
2269 default_recv_ssrc_ = ssrc; | |
2270 SetOutputVolume(default_recv_ssrc_, default_recv_volume_); | |
2271 if (default_sink_) { | |
2272 std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( | |
2273 new ProxySink(default_sink_.get())); | |
2274 SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink)); | |
2275 } | |
2276 } | 2274 } |
2277 | 2275 |
2278 // Forward packet to Call. If the SSRC is unknown we'll return after this. | 2276 StreamParams sp; |
2279 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, | 2277 sp.ssrcs.push_back(ssrc); |
2280 packet_time.not_before); | 2278 LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; |
2281 webrtc::PacketReceiver::DeliveryStatus delivery_result = | 2279 if (!AddRecvStream(sp)) { |
2282 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, | 2280 LOG(LS_WARNING) << "Could not create default receive stream."; |
2283 packet->cdata(), packet->size(), webrtc_packet_time); | 2281 return; |
2284 if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) { | |
2285 // If the SSRC is unknown here, route it to the default channel, if we have | |
2286 // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208 | |
2287 if (default_recv_ssrc_ == -1) { | |
2288 return; | |
2289 } else { | |
2290 ssrc = default_recv_ssrc_; | |
2291 } | |
2292 } | 2282 } |
2293 | 2283 default_recv_ssrc_ = ssrc; |
2294 // Find the channel to send this packet to. It must exist since webrtc::Call | 2284 SetOutputVolume(default_recv_ssrc_, default_recv_volume_); |
2295 // was able to demux the packet. | 2285 if (default_sink_) { |
2296 int channel = GetReceiveChannelId(ssrc); | 2286 std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink( |
2297 RTC_DCHECK(channel != -1); | 2287 new ProxySink(default_sink_.get())); |
2298 | 2288 SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink)); |
2299 // Pass it off to the decoder. | 2289 } |
2300 engine()->voe()->network()->ReceivedRTPPacket( | 2290 delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
2301 channel, packet->cdata(), packet->size(), webrtc_packet_time); | 2291 packet->cdata(), |
2292 packet->size(), | |
2293 webrtc_packet_time); | |
2294 RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result); | |
2302 } | 2295 } |
2303 | 2296 |
2304 void WebRtcVoiceMediaChannel::OnRtcpReceived( | 2297 void WebRtcVoiceMediaChannel::OnRtcpReceived( |
2305 rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { | 2298 rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { |
2306 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2299 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
2307 | 2300 |
2308 // Forward packet to Call as well. | 2301 // Forward packet to Call as well. |
2309 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, | 2302 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
2310 packet_time.not_before); | 2303 packet_time.not_before); |
2311 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, | 2304 call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, |
2312 packet->cdata(), packet->size(), webrtc_packet_time); | 2305 packet->cdata(), packet->size(), webrtc_packet_time); |
2313 | |
2314 // Sending channels need all RTCP packets with feedback information. | |
2315 // Even sender reports can contain attached report blocks. | |
2316 // Receiving channels need sender reports in order to create | |
2317 // correct receiver reports. | |
2318 int type = 0; | |
2319 if (!GetRtcpType(packet->cdata(), packet->size(), &type)) { | |
2320 LOG(LS_WARNING) << "Failed to parse type from received RTCP packet"; | |
2321 return; | |
2322 } | |
2323 | |
2324 // If it is a sender report, find the receive channel that is listening. | |
2325 if (type == kRtcpTypeSR) { | |
2326 uint32_t ssrc = 0; | |
2327 if (!GetRtcpSsrc(packet->cdata(), packet->size(), &ssrc)) { | |
2328 return; | |
2329 } | |
2330 int recv_channel_id = GetReceiveChannelId(ssrc); | |
2331 if (recv_channel_id != -1) { | |
2332 engine()->voe()->network()->ReceivedRTCPPacket( | |
2333 recv_channel_id, packet->cdata(), packet->size()); | |
2334 } | |
2335 } | |
2336 | |
2337 // SR may continue RR and any RR entry may correspond to any one of the send | |
2338 // channels. So all RTCP packets must be forwarded all send channels. VoE | |
2339 // will filter out RR internally. | |
2340 for (const auto& ch : send_streams_) { | |
2341 engine()->voe()->network()->ReceivedRTCPPacket( | |
2342 ch.second->channel(), packet->cdata(), packet->size()); | |
2343 } | |
2344 } | 2306 } |
2345 | 2307 |
2346 void WebRtcVoiceMediaChannel::OnNetworkRouteChanged( | 2308 void WebRtcVoiceMediaChannel::OnNetworkRouteChanged( |
2347 const std::string& transport_name, | 2309 const std::string& transport_name, |
2348 const rtc::NetworkRoute& network_route) { | 2310 const rtc::NetworkRoute& network_route) { |
2349 call_->OnNetworkRouteChanged(transport_name, network_route); | 2311 call_->OnNetworkRouteChanged(transport_name, network_route); |
2350 } | 2312 } |
2351 | 2313 |
2352 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { | 2314 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { |
2353 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2315 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
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2575 } | 2537 } |
2576 } else { | 2538 } else { |
2577 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2539 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
2578 engine()->voe()->base()->StopPlayout(channel); | 2540 engine()->voe()->base()->StopPlayout(channel); |
2579 } | 2541 } |
2580 return true; | 2542 return true; |
2581 } | 2543 } |
2582 } // namespace cricket | 2544 } // namespace cricket |
2583 | 2545 |
2584 #endif // HAVE_WEBRTC_VOICE | 2546 #endif // HAVE_WEBRTC_VOICE |
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