aboutsummaryrefslogtreecommitdiff
path: root/talk/media/webrtc/webrtcvoiceengine.h
diff options
context:
space:
mode:
Diffstat (limited to 'talk/media/webrtc/webrtcvoiceengine.h')
-rw-r--r--talk/media/webrtc/webrtcvoiceengine.h145
1 files changed, 43 insertions, 102 deletions
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 1cf05e71a2..0f2f59e492 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -29,7 +29,6 @@
#define TALK_MEDIA_WEBRTCVOICEENGINE_H_
#include <map>
-#include <set>
#include <string>
#include <vector>
@@ -37,9 +36,8 @@
#include "talk/media/webrtc/webrtccommon.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "talk/session/media/channel.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/buffer.h"
-#include "webrtc/base/byteorder.h"
-#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/thread_checker.h"
@@ -51,43 +49,34 @@ namespace cricket {
class AudioDeviceModule;
class AudioRenderer;
-class VoETraceWrapper;
class VoEWrapper;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
// It uses the WebRtc VoiceEngine library for audio handling.
-class WebRtcVoiceEngine
- : public webrtc::VoiceEngineObserver,
- public webrtc::TraceCallback {
+class WebRtcVoiceEngine final : public webrtc::TraceCallback {
friend class WebRtcVoiceMediaChannel;
-
public:
+ // Exposed for the WVoE/MC unit test.
+ static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
+
WebRtcVoiceEngine();
// Dependency injection for testing.
- WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
+ explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
~WebRtcVoiceEngine();
bool Init(rtc::Thread* worker_thread);
void Terminate();
- webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
+ rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const AudioOptions& options);
- AudioOptions GetOptions() const { return options_; }
- bool SetOptions(const AudioOptions& options);
- bool SetDevices(const Device* in_device, const Device* out_device);
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
int GetInputLevel();
const std::vector<AudioCodec>& codecs();
- bool FindCodec(const AudioCodec& codec);
- bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
-
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
-
- void SetLogging(int min_sev, const char* filter);
+ RtpCapabilities GetCapabilities() const;
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
@@ -120,68 +109,49 @@ class WebRtcVoiceEngine
private:
void Construct();
- void ConstructCodecs();
- bool GetVoeCodec(int index, webrtc::CodecInst* codec);
bool InitInternal();
- void SetTraceFilter(int filter);
- void SetTraceOptions(const std::string& options);
// Every option that is "set" will be applied. Every option not "set" will be
// ignored. This allows us to selectively turn on and off different options
// easily at any time.
bool ApplyOptions(const AudioOptions& options);
+ void SetDefaultDevices();
// webrtc::TraceCallback:
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
- // webrtc::VoiceEngineObserver:
- void CallbackOnError(int channel_id, int errCode) override;
-
- // Given the device type, name, and id, find device id. Return true and
- // set the output parameter rtc_id if successful.
- bool FindWebRtcAudioDeviceId(
- bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
-
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
- static const int kDefaultLogSeverity = rtc::LS_WARNING;
+ rtc::ThreadChecker signal_thread_checker_;
+ rtc::ThreadChecker worker_thread_checker_;
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
- rtc::scoped_ptr<VoETraceWrapper> tracing_;
+ rtc::scoped_refptr<webrtc::AudioState> audio_state_;
// The external audio device manager
- webrtc::AudioDeviceModule* adm_;
- int log_filter_;
- std::string log_options_;
- bool is_dumping_aec_;
+ webrtc::AudioDeviceModule* adm_ = nullptr;
std::vector<AudioCodec> codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
- // channels_ can be read from WebRtc callback thread. We need a lock on that
- // callback as well as the RegisterChannel/UnregisterChannel.
- rtc::CriticalSection channels_cs_;
- webrtc::AgcConfig default_agc_config_;
-
webrtc::Config voe_config_;
+ bool initialized_ = false;
+ bool is_dumping_aec_ = false;
- bool initialized_;
- AudioOptions options_;
-
+ webrtc::AgcConfig default_agc_config_;
// Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
// values, and apply them in case they are missing in the audio options. We
// need to do this because SetExtraOptions() will revert to defaults for
// options which are not provided.
- Settable<bool> extended_filter_aec_;
- Settable<bool> delay_agnostic_aec_;
- Settable<bool> experimental_ns_;
+ rtc::Optional<bool> extended_filter_aec_;
+ rtc::Optional<bool> delay_agnostic_aec_;
+ rtc::Optional<bool> experimental_ns_;
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
};
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
// WebRtc Voice Engine.
-class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
- public webrtc::Transport {
+class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
+ public webrtc::Transport {
public:
WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const AudioOptions& options,
@@ -217,7 +187,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool CanInsertDtmf() override;
- bool InsertDtmf(uint32_t ssrc, int event, int duration, int flags) override;
+ bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
void OnPacketReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
@@ -226,6 +196,10 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
void OnReadyToSend(bool ready) override {}
bool GetStats(VoiceMediaInfo* info) override;
+ void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+
// implements Transport interface
bool SendRtp(const uint8_t* data,
size_t len,
@@ -243,20 +217,14 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
}
- void OnError(int error);
-
int GetReceiveChannelId(uint32_t ssrc) const;
int GetSendChannelId(uint32_t ssrc) const;
private:
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
- bool SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
bool SetOptions(const AudioOptions& options);
bool SetMaxSendBandwidth(int bps);
bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
- bool SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
bool MuteStream(uint32_t ssrc, bool mute);
@@ -267,82 +235,55 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
const std::vector<AudioCodec>& all_codecs,
webrtc::CodecInst* send_codec);
bool SetPlayout(int channel, bool playout);
- static Error WebRtcErrorToChannelError(int err_code);
-
- typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
- unsigned char);
-
void SetNack(int channel, bool nack_enabled);
bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
bool ChangePlayout(bool playout);
bool ChangeSend(SendFlags send);
bool ChangeSend(int channel, SendFlags send);
- bool ConfigureRecvChannel(int channel);
int CreateVoEChannel();
- bool DeleteChannel(int channel);
+ bool DeleteVoEChannel(int channel);
bool IsDefaultRecvStream(uint32_t ssrc) {
return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
bool SetSendBitrateInternal(int bps);
- bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
- const RtpHeaderExtension* extension);
- void RecreateAudioReceiveStreams();
- void AddAudioReceiveStream(uint32_t ssrc);
- void RemoveAudioReceiveStream(uint32_t ssrc);
- bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
-
- bool SetChannelRecvRtpHeaderExtensions(
- int channel_id,
- const std::vector<RtpHeaderExtension>& extensions);
- bool SetChannelSendRtpHeaderExtensions(
- int channel_id,
- const std::vector<RtpHeaderExtension>& extensions);
+ rtc::ThreadChecker worker_thread_checker_;
- rtc::ThreadChecker thread_checker_;
-
- WebRtcVoiceEngine* const engine_;
+ WebRtcVoiceEngine* const engine_ = nullptr;
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
- bool send_bitrate_setting_;
- int send_bitrate_bps_;
+ bool send_bitrate_setting_ = false;
+ int send_bitrate_bps_ = 0;
AudioOptions options_;
- bool dtmf_allowed_;
- bool desired_playout_;
- bool nack_enabled_;
- bool playout_;
- bool typing_noise_detected_;
- SendFlags desired_send_;
- SendFlags send_;
- webrtc::Call* const call_;
+ rtc::Optional<int> dtmf_payload_type_;
+ bool desired_playout_ = false;
+ bool nack_enabled_ = false;
+ bool playout_ = false;
+ SendFlags desired_send_ = SEND_NOTHING;
+ SendFlags send_ = SEND_NOTHING;
+ webrtc::Call* const call_ = nullptr;
// SSRC of unsignalled receive stream, or -1 if there isn't one.
int64_t default_recv_ssrc_ = -1;
// Volume for unsignalled stream, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
- // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
+ // Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
- uint32_t receiver_reports_ssrc_ = 1;
+ // and https://code.google.com/p/chromium/issues/detail?id=547661
+ uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
- std::vector<RtpHeaderExtension> send_extensions_;
+ std::vector<webrtc::RtpExtension> send_rtp_extensions_;
class WebRtcAudioReceiveStream;
- std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_;
- std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
- std::map<uint32_t, StreamParams> receive_stream_params_;
- // receive_channels_ can be read from WebRtc callback thread. Access from
- // the WebRtc thread must be synchronized with edits on the worker thread.
- // Reads on the worker thread are ok.
- std::vector<RtpHeaderExtension> receive_extensions_;
+ std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
};
-
} // namespace cricket
#endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-19 00:56:18 +0000