blob: 5266f55550cfe0537eb9dc88210168a935da5fd4 [file] [log] [blame]
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_MEDIA_WEBRTCVOICEENGINE_H_
29#define TALK_MEDIA_WEBRTCVOICEENGINE_H_
30
31#include <map>
32#include <set>
33#include <string>
34#include <vector>
35
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036#include "talk/media/base/rtputils.h"
37#include "talk/media/webrtc/webrtccommon.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/media/webrtc/webrtcvoe.h"
39#include "talk/session/media/channel.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000040#include "webrtc/base/buffer.h"
41#include "webrtc/base/byteorder.h"
42#include "webrtc/base/logging.h"
43#include "webrtc/base/scoped_ptr.h"
44#include "webrtc/base/stream.h"
Fredrik Solenberg4b60c732015-05-07 14:07:48 +020045#include "webrtc/base/thread_checker.h"
46#include "webrtc/call.h"
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +000047#include "webrtc/common.h"
Henrik Lundin64dad832015-05-11 12:44:23 +020048#include "webrtc/config.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050namespace cricket {
51
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052class AudioDeviceModule;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000053class AudioRenderer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054class VoETraceWrapper;
55class VoEWrapper;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056class WebRtcVoiceMediaChannel;
57
58// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
59// It uses the WebRtc VoiceEngine library for audio handling.
60class WebRtcVoiceEngine
61 : public webrtc::VoiceEngineObserver,
Fredrik Solenberg7d173362015-09-23 12:23:21 +020062 public webrtc::TraceCallback {
Jelena Marusicc28a8962015-05-29 15:05:44 +020063 friend class WebRtcVoiceMediaChannel;
64
henrike@webrtc.org28e20752013-07-10 00:45:36 +000065 public:
66 WebRtcVoiceEngine();
67 // Dependency injection for testing.
Fredrik Solenbergccb49e72015-05-19 11:37:56 +020068 WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000069 ~WebRtcVoiceEngine();
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000070 bool Init(rtc::Thread* worker_thread);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071 void Terminate();
72
Fredrik Solenberg709ed672015-09-15 12:26:33 +020073 webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
74 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
75 const AudioOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +000077 AudioOptions GetOptions() const { return options_; }
78 bool SetOptions(const AudioOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079 bool SetDevices(const Device* in_device, const Device* out_device);
80 bool GetOutputVolume(int* level);
81 bool SetOutputVolume(int level);
82 int GetInputLevel();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000083
84 const std::vector<AudioCodec>& codecs();
85 bool FindCodec(const AudioCodec& codec);
86 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
87
88 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
89
90 void SetLogging(int min_sev, const char* filter);
91
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092 // For tracking WebRtc channels. Needed because we have to pause them
93 // all when switching devices.
94 // May only be called by WebRtcVoiceMediaChannel.
95 void RegisterChannel(WebRtcVoiceMediaChannel *channel);
96 void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
97
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098 // Called by WebRtcVoiceMediaChannel to set a gain offset from
99 // the default AGC target level.
100 bool AdjustAgcLevel(int delta);
101
102 VoEWrapper* voe() { return voe_wrapper_.get(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 int GetLastEngineError();
104
Fredrik Solenbergccb49e72015-05-19 11:37:56 +0200105 // Set the external ADM. This can only be called before Init.
106 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107
wu@webrtc.orga9890802013-12-13 00:21:03 +0000108 // Starts AEC dump using existing file.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000109 bool StartAecDump(rtc::PlatformFile file);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000110
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000111 // Create a VoiceEngine Channel.
112 int CreateMediaVoiceChannel();
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000113
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000114 private:
Fredrik Solenbergaf9fb212015-08-26 10:45:53 +0200115 typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116
117 void Construct();
118 void ConstructCodecs();
henrik.lundin@webrtc.org8038d422014-11-11 08:38:24 +0000119 bool GetVoeCodec(int index, webrtc::CodecInst* codec);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120 bool InitInternal();
121 void SetTraceFilter(int filter);
122 void SetTraceOptions(const std::string& options);
123 // Applies either options or overrides. Every option that is "set"
124 // will be applied. Every option not "set" will be ignored. This
125 // allows us to selectively turn on and off different options easily
126 // at any time.
127 bool ApplyOptions(const AudioOptions& options);
Jelena Marusicc28a8962015-05-29 15:05:44 +0200128 // Overrides, when set, take precedence over the options on a
129 // per-option basis. For example, if AGC is set in options and AEC
130 // is set in overrides, AGC and AEC will be both be set. Overrides
131 // can also turn off options. For example, if AGC is set to "on" in
132 // options and AGC is set to "off" in overrides, the result is that
133 // AGC will be off until different overrides are applied or until
134 // the overrides are cleared. Only one set of overrides is present
135 // at a time (they do not "stack"). And when the overrides are
136 // cleared, the media engine's state reverts back to the options set
137 // via SetOptions. This allows us to have both "persistent options"
138 // (the normal options) and "temporary options" (overrides).
139 bool SetOptionOverrides(const AudioOptions& options);
140 bool ClearOptionOverrides();
xians@webrtc.org3cefbc92014-10-10 09:42:53 +0000141
142 // webrtc::TraceCallback:
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000143 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
xians@webrtc.org3cefbc92014-10-10 09:42:53 +0000144
145 // webrtc::VoiceEngineObserver:
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000146 void CallbackOnError(int channel, int errCode) override;
xians@webrtc.org3cefbc92014-10-10 09:42:53 +0000147
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000148 // Given the device type, name, and id, find device id. Return true and
149 // set the output parameter rtc_id if successful.
150 bool FindWebRtcAudioDeviceId(
151 bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
152 bool FindChannelAndSsrc(int channel_num,
153 WebRtcVoiceMediaChannel** channel,
154 uint32* ssrc) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155
156 void StartAecDump(const std::string& filename);
157 void StopAecDump();
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000158 int CreateVoiceChannel(VoEWrapper* voe);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000159
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000160 static const int kDefaultLogSeverity = rtc::LS_WARNING;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000161
162 // The primary instance of WebRtc VoiceEngine.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000163 rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000164 rtc::scoped_ptr<VoETraceWrapper> tracing_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165 // The external audio device manager
166 webrtc::AudioDeviceModule* adm_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000167 int log_filter_;
168 std::string log_options_;
169 bool is_dumping_aec_;
170 std::vector<AudioCodec> codecs_;
171 std::vector<RtpHeaderExtension> rtp_header_extensions_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172 ChannelList channels_;
173 // channels_ can be read from WebRtc callback thread. We need a lock on that
174 // callback as well as the RegisterChannel/UnregisterChannel.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000175 rtc::CriticalSection channels_cs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000176 webrtc::AgcConfig default_agc_config_;
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000177
178 webrtc::Config voe_config_;
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000179
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000180 bool initialized_;
181 // See SetOptions and SetOptionOverrides for a description of the
182 // difference between options and overrides.
183 // options_ are the base options, which combined with the
184 // option_overrides_, create the current options being used.
185 // options_ is stored so that when option_overrides_ is cleared, we
186 // can restore the options_ without the option_overrides.
187 AudioOptions options_;
188 AudioOptions option_overrides_;
189
Henrik Lundin441f6342015-06-09 16:03:13 +0200190 // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
Bjorn Volckerbf395c12015-03-25 22:45:56 +0100191 // values, and apply them in case they are missing in the audio options. We
192 // need to do this because SetExtraOptions() will revert to defaults for
193 // options which are not provided.
Henrik Lundin441f6342015-06-09 16:03:13 +0200194 Settable<bool> extended_filter_aec_;
Bjorn Volckerbf395c12015-03-25 22:45:56 +0100195 Settable<bool> delay_agnostic_aec_;
buildbot@webrtc.org1f8a2372014-08-28 10:52:44 +0000196 Settable<bool> experimental_ns_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197};
198
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000199// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
200// WebRtc Voice Engine.
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200201class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
202 public webrtc::Transport {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000203 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200204 WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
205 const AudioOptions& options,
206 webrtc::Call* call);
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200207 ~WebRtcVoiceMediaChannel() override;
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200208
209 int voe_channel() const { return voe_channel_; }
210 bool valid() const { return voe_channel_ != -1; }
solenberg66f43392015-09-09 01:36:22 -0700211 const AudioOptions& options() const { return options_; }
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200212
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700213 bool SetSendParameters(const AudioSendParameters& params) override;
214 bool SetRecvParameters(const AudioRecvParameters& params) override;
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200215 bool SetPlayout(bool playout) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000216 bool PausePlayout();
217 bool ResumePlayout();
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200218 bool SetSend(SendFlags send) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000219 bool PauseSend();
220 bool ResumeSend();
solenberg1dd98f32015-09-10 01:57:14 -0700221 bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
222 AudioRenderer* renderer) override;
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200223 bool AddSendStream(const StreamParams& sp) override;
224 bool RemoveSendStream(uint32 ssrc) override;
225 bool AddRecvStream(const StreamParams& sp) override;
226 bool RemoveRecvStream(uint32 ssrc) override;
227 bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200228 bool GetActiveStreams(AudioInfo::StreamList* actives) override;
229 int GetOutputLevel() override;
230 int GetTimeSinceLastTyping() override;
231 void SetTypingDetectionParameters(int time_window,
232 int cost_per_typing,
233 int reporting_threshold,
234 int penalty_decay,
235 int type_event_delay) override;
236 bool SetOutputScaling(uint32 ssrc, double left, double right) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000237
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200238 bool CanInsertDtmf() override;
239 bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000240
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200241 void OnPacketReceived(rtc::Buffer* packet,
242 const rtc::PacketTime& packet_time) override;
243 void OnRtcpReceived(rtc::Buffer* packet,
244 const rtc::PacketTime& packet_time) override;
245 void OnReadyToSend(bool ready) override {}
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200246 bool GetStats(VoiceMediaInfo* info) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 // Gets last reported error from WebRtc voice engine. This should be only
248 // called in response a failure.
Fredrik Solenbergaaf8ff22015-05-07 16:05:53 +0200249 void GetLastMediaError(uint32* ssrc,
250 VoiceMediaChannel::Error* error) override;
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200251
252 // implements Transport interface
pbos2d566682015-09-28 09:59:31 -0700253 bool SendRtp(const uint8_t* data, size_t len) override {
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200254 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
255 kMaxRtpPacketLen);
pbos2d566682015-09-28 09:59:31 -0700256 return VoiceMediaChannel::SendPacket(&packet);
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200257 }
258
pbos2d566682015-09-28 09:59:31 -0700259 bool SendRtcp(const uint8_t* data, size_t len) override {
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200260 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
261 kMaxRtpPacketLen);
pbos2d566682015-09-28 09:59:31 -0700262 return VoiceMediaChannel::SendRtcp(&packet);
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200263 }
264
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000265 bool FindSsrc(int channel_num, uint32* ssrc);
266 void OnError(uint32 ssrc, int error);
267
268 bool sending() const { return send_ != SEND_NOTHING; }
Fredrik Solenbergaf9fb212015-08-26 10:45:53 +0200269 int GetReceiveChannelNum(uint32 ssrc) const;
270 int GetSendChannelNum(uint32 ssrc) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200272 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200273 bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
274 bool SetSendRtpHeaderExtensions(
275 const std::vector<RtpHeaderExtension>& extensions);
276 bool SetOptions(const AudioOptions& options);
277 bool SetMaxSendBandwidth(int bps);
278 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
279 bool SetRecvRtpHeaderExtensions(
280 const std::vector<RtpHeaderExtension>& extensions);
solenberg1dd98f32015-09-10 01:57:14 -0700281 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
282 bool MuteStream(uint32 ssrc, bool mute);
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200283
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200284 WebRtcVoiceEngine* engine() { return engine_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000285 int GetLastEngineError() { return engine()->GetLastEngineError(); }
286 int GetOutputLevel(int channel);
287 bool GetRedSendCodec(const AudioCodec& red_codec,
288 const std::vector<AudioCodec>& all_codecs,
289 webrtc::CodecInst* send_codec);
290 bool EnableRtcp(int channel);
291 bool ResetRecvCodecs(int channel);
292 bool SetPlayout(int channel, bool playout);
293 static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
294 static Error WebRtcErrorToChannelError(int err_code);
295
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000296 class WebRtcVoiceChannelRenderer;
297 // Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of
298 // WebRtcVoiceChannelRenderer will be created for every new stream and
299 // will be destroyed when the stream goes away.
300 typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap;
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000301 typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
302 unsigned char);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000303
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000304 void SetNack(int channel, bool nack_enabled);
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000305 void SetNack(const ChannelMap& channels, bool nack_enabled);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000306 bool SetSendCodec(const webrtc::CodecInst& send_codec);
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000307 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000308 bool ChangePlayout(bool playout);
309 bool ChangeSend(SendFlags send);
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000310 bool ChangeSend(int channel, SendFlags send);
311 void ConfigureSendChannel(int channel);
wu@webrtc.org78187522013-10-07 23:32:02 +0000312 bool ConfigureRecvChannel(int channel);
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000313 bool DeleteChannel(int channel);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000314 bool InConferenceMode() const {
315 return options_.conference_mode.GetWithDefaultIfUnset(false);
316 }
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000317 bool IsDefaultChannel(int channel_id) const {
318 return channel_id == voe_channel();
319 }
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000320 bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
minyue@webrtc.org26236952014-10-29 02:27:08 +0000321 bool SetSendBitrateInternal(int bps);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000322
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000323 bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
324 const RtpHeaderExtension* extension);
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200325 void RecreateAudioReceiveStreams();
326 void AddAudioReceiveStream(uint32 ssrc);
327 void RemoveAudioReceiveStream(uint32 ssrc);
Fredrik Solenbergaf9fb212015-08-26 10:45:53 +0200328 bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200329
buildbot@webrtc.org150835e2014-05-06 15:54:38 +0000330 bool SetChannelRecvRtpHeaderExtensions(
331 int channel_id,
332 const std::vector<RtpHeaderExtension>& extensions);
333 bool SetChannelSendRtpHeaderExtensions(
334 int channel_id,
335 const std::vector<RtpHeaderExtension>& extensions);
336
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200337 rtc::ThreadChecker thread_checker_;
338
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200339 WebRtcVoiceEngine* const engine_;
Fredrik Solenberge444a3d2015-05-07 16:42:08 +0200340 const int voe_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000341 std::vector<AudioCodec> recv_codecs_;
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000342 std::vector<AudioCodec> send_codecs_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000343 rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
minyue@webrtc.org26236952014-10-29 02:27:08 +0000344 bool send_bitrate_setting_;
345 int send_bitrate_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000346 AudioOptions options_;
347 bool dtmf_allowed_;
348 bool desired_playout_;
349 bool nack_enabled_;
350 bool playout_;
wu@webrtc.org967bfff2013-09-19 05:49:50 +0000351 bool typing_noise_detected_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352 SendFlags desired_send_;
353 SendFlags send_;
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200354 webrtc::Call* const call_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000355
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000356 // send_channels_ contains the channels which are being used for sending.
357 // When the default channel (voe_channel) is used for sending, it is
358 // contained in send_channels_, otherwise not.
359 ChannelMap send_channels_;
buildbot@webrtc.org150835e2014-05-06 15:54:38 +0000360 std::vector<RtpHeaderExtension> send_extensions_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000361 uint32 default_receive_ssrc_;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000362 // Note the default channel (voe_channel()) can reside in both
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000363 // receive_channels_ and send_channels_ in non-conference mode and in that
364 // case it will only be there if a non-zero default_receive_ssrc_ is set.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000365 ChannelMap receive_channels_; // for multiple sources
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200366 std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_;
pbos8fc7fa72015-07-15 08:02:58 -0700367 std::map<uint32, StreamParams> receive_stream_params_;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000368 // receive_channels_ can be read from WebRtc callback thread. Access from
369 // the WebRtc thread must be synchronized with edits on the worker thread.
370 // Reads on the worker thread are ok.
buildbot@webrtc.org150835e2014-05-06 15:54:38 +0000371 std::vector<RtpHeaderExtension> receive_extensions_;
Fredrik Solenberg4b60c732015-05-07 14:07:48 +0200372 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
373
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000374 // Do not lock this on the VoE media processor thread; potential for deadlock
375 // exists.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000376 mutable rtc::CriticalSection receive_channels_cs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000377};
378
379} // namespace cricket
380
381#endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_