- Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel.
- Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer - Create webrtc::AudioSendStreams. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1415563003 Cr-Commit-Position: refs/heads/master@{#10361}
This commit is contained in:
solenberg
Commit bot
parent
f4d23b2254
commit
c96df779b0
@ -34,6 +34,20 @@
|
||||
#include "webrtc/base/gunit.h"
|
||||
|
||||
namespace cricket {
|
||||
FakeAudioSendStream::FakeAudioSendStream(
|
||||
const webrtc::AudioSendStream::Config& config) : config_(config) {
|
||||
RTC_DCHECK(config.voe_channel_id != -1);
|
||||
}
|
||||
|
||||
webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
|
||||
return webrtc::AudioSendStream::Stats();
|
||||
}
|
||||
|
||||
const webrtc::AudioSendStream::Config&
|
||||
FakeAudioSendStream::GetConfig() const {
|
||||
return config_;
|
||||
}
|
||||
|
||||
FakeAudioReceiveStream::FakeAudioReceiveStream(
|
||||
const webrtc::AudioReceiveStream::Config& config)
|
||||
: config_(config), received_packets_(0) {
|
||||
@ -206,6 +220,7 @@ FakeCall::FakeCall(const webrtc::Call::Config& config)
|
||||
|
||||
FakeCall::~FakeCall() {
|
||||
EXPECT_EQ(0u, video_send_streams_.size());
|
||||
EXPECT_EQ(0u, audio_send_streams_.size());
|
||||
EXPECT_EQ(0u, video_receive_streams_.size());
|
||||
EXPECT_EQ(0u, audio_receive_streams_.size());
|
||||
}
|
||||
@ -222,12 +237,25 @@ const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
|
||||
return video_receive_streams_;
|
||||
}
|
||||
|
||||
const std::vector<FakeAudioSendStream*>& FakeCall::GetAudioSendStreams() {
|
||||
return audio_send_streams_;
|
||||
}
|
||||
|
||||
const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) {
|
||||
for (const auto* p : GetAudioSendStreams()) {
|
||||
if (p->GetConfig().rtp.ssrc == ssrc) {
|
||||
return p;
|
||||
}
|
||||
}
|
||||
return nullptr;
|
||||
}
|
||||
|
||||
const std::vector<FakeAudioReceiveStream*>& FakeCall::GetAudioReceiveStreams() {
|
||||
return audio_receive_streams_;
|
||||
}
|
||||
|
||||
const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
|
||||
for (const auto p : GetAudioReceiveStreams()) {
|
||||
for (const auto* p : GetAudioReceiveStreams()) {
|
||||
if (p->GetConfig().rtp.remote_ssrc == ssrc) {
|
||||
return p;
|
||||
}
|
||||
@ -241,10 +269,22 @@ webrtc::NetworkState FakeCall::GetNetworkState() const {
|
||||
|
||||
webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
|
||||
const webrtc::AudioSendStream::Config& config) {
|
||||
return nullptr;
|
||||
FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config);
|
||||
audio_send_streams_.push_back(fake_stream);
|
||||
++num_created_send_streams_;
|
||||
return fake_stream;
|
||||
}
|
||||
|
||||
void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
|
||||
auto it = std::find(audio_send_streams_.begin(),
|
||||
audio_send_streams_.end(),
|
||||
static_cast<FakeAudioSendStream*>(send_stream));
|
||||
if (it == audio_send_streams_.end()) {
|
||||
ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter.";
|
||||
} else {
|
||||
delete *it;
|
||||
audio_send_streams_.erase(it);
|
||||
}
|
||||
}
|
||||
|
||||
webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
|
||||
|
||||
@ -25,6 +25,15 @@
|
||||
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*/
|
||||
|
||||
// This file contains fake implementations, for use in unit tests, of the
|
||||
// following classes:
|
||||
//
|
||||
// webrtc::Call
|
||||
// webrtc::AudioSendStream
|
||||
// webrtc::AudioReceiveStream
|
||||
// webrtc::VideoSendStream
|
||||
// webrtc::VideoReceiveStream
|
||||
|
||||
#ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
|
||||
#define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
|
||||
|
||||
@ -32,11 +41,35 @@
|
||||
|
||||
#include "webrtc/call.h"
|
||||
#include "webrtc/audio_receive_stream.h"
|
||||
#include "webrtc/audio_send_stream.h"
|
||||
#include "webrtc/video_frame.h"
|
||||
#include "webrtc/video_receive_stream.h"
|
||||
#include "webrtc/video_send_stream.h"
|
||||
|
||||
namespace cricket {
|
||||
|
||||
class FakeAudioSendStream : public webrtc::AudioSendStream {
|
||||
public:
|
||||
explicit FakeAudioSendStream(
|
||||
const webrtc::AudioSendStream::Config& config);
|
||||
|
||||
// webrtc::AudioSendStream implementation.
|
||||
webrtc::AudioSendStream::Stats GetStats() const override;
|
||||
|
||||
const webrtc::AudioSendStream::Config& GetConfig() const;
|
||||
|
||||
private:
|
||||
// webrtc::SendStream implementation.
|
||||
void Start() override {}
|
||||
void Stop() override {}
|
||||
void SignalNetworkState(webrtc::NetworkState state) override {}
|
||||
bool DeliverRtcp(const uint8_t* packet, size_t length) override {
|
||||
return true;
|
||||
}
|
||||
|
||||
webrtc::AudioSendStream::Config config_;
|
||||
};
|
||||
|
||||
class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
|
||||
public:
|
||||
explicit FakeAudioReceiveStream(
|
||||
@ -161,6 +194,8 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
|
||||
const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
|
||||
const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
|
||||
|
||||
const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
|
||||
const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
|
||||
const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
|
||||
const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
|
||||
|
||||
@ -208,6 +243,7 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
|
||||
rtc::SentPacket last_sent_packet_;
|
||||
webrtc::Call::Stats stats_;
|
||||
std::vector<FakeVideoSendStream*> video_send_streams_;
|
||||
std::vector<FakeAudioSendStream*> audio_send_streams_;
|
||||
std::vector<FakeVideoReceiveStream*> video_receive_streams_;
|
||||
std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
|
||||
|
||||
|
||||
@ -1313,15 +1313,25 @@ int WebRtcVoiceEngine::CreateVoEChannel() {
|
||||
return voe_wrapper_->base()->CreateChannel(voe_config_);
|
||||
}
|
||||
|
||||
class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
||||
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
|
||||
: public AudioRenderer::Sink {
|
||||
public:
|
||||
WebRtcVoiceChannelRenderer(int ch,
|
||||
webrtc::AudioTransport* voe_audio_transport)
|
||||
WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
|
||||
uint32_t ssrc, webrtc::Call* call)
|
||||
: channel_(ch),
|
||||
voe_audio_transport_(voe_audio_transport),
|
||||
renderer_(NULL) {}
|
||||
~WebRtcVoiceChannelRenderer() override { Stop(); }
|
||||
call_(call) {
|
||||
RTC_DCHECK(call);
|
||||
webrtc::AudioSendStream::Config config(nullptr);
|
||||
config.voe_channel_id = channel_;
|
||||
config.rtp.ssrc = ssrc;
|
||||
stream_ = call_->CreateAudioSendStream(config);
|
||||
RTC_DCHECK(stream_);
|
||||
}
|
||||
~WebRtcAudioSendStream() override {
|
||||
Stop();
|
||||
call_->DestroyAudioSendStream(stream_);
|
||||
}
|
||||
|
||||
// Starts the rendering by setting a sink to the renderer to get data
|
||||
// callback.
|
||||
@ -1329,8 +1339,8 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
||||
// TODO(xians): Make sure Start() is called only once.
|
||||
void Start(AudioRenderer* renderer) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
RTC_DCHECK(renderer != NULL);
|
||||
if (renderer_ != NULL) {
|
||||
RTC_DCHECK(renderer);
|
||||
if (renderer_) {
|
||||
RTC_DCHECK(renderer_ == renderer);
|
||||
return;
|
||||
}
|
||||
@ -1338,14 +1348,14 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
||||
renderer_ = renderer;
|
||||
}
|
||||
|
||||
// Stops rendering by setting the sink of the renderer to NULL. No data
|
||||
// Stops rendering by setting the sink of the renderer to nullptr. No data
|
||||
// callback will be received after this method.
|
||||
// This method is called on the libjingle worker thread.
|
||||
void Stop() {
|
||||
rtc::CritScope lock(&lock_);
|
||||
if (renderer_ != NULL) {
|
||||
renderer_->SetSink(NULL);
|
||||
renderer_ = NULL;
|
||||
if (renderer_) {
|
||||
renderer_->SetSink(nullptr);
|
||||
renderer_ = nullptr;
|
||||
}
|
||||
}
|
||||
|
||||
@ -1356,6 +1366,7 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
||||
int sample_rate,
|
||||
int number_of_channels,
|
||||
size_t number_of_frames) override {
|
||||
RTC_DCHECK(voe_audio_transport_);
|
||||
voe_audio_transport_->OnData(channel_,
|
||||
audio_data,
|
||||
bits_per_sample,
|
||||
@ -1368,25 +1379,42 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
|
||||
// never been called, this callback won't be triggered.
|
||||
void OnClose() override {
|
||||
rtc::CritScope lock(&lock_);
|
||||
// Set |renderer_| to NULL to make sure no more callback will get into
|
||||
// Set |renderer_| to nullptr to make sure no more callback will get into
|
||||
// the renderer.
|
||||
renderer_ = NULL;
|
||||
renderer_ = nullptr;
|
||||
}
|
||||
|
||||
// Accessor to the VoE channel ID.
|
||||
int channel() const { return channel_; }
|
||||
|
||||
private:
|
||||
const int channel_;
|
||||
webrtc::AudioTransport* const voe_audio_transport_;
|
||||
const int channel_ = -1;
|
||||
webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
|
||||
webrtc::Call* call_ = nullptr;
|
||||
webrtc::AudioSendStream* stream_ = nullptr;
|
||||
|
||||
// Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
|
||||
// PeerConnection will make sure invalidating the pointer before the object
|
||||
// goes away.
|
||||
AudioRenderer* renderer_;
|
||||
AudioRenderer* renderer_ = nullptr;
|
||||
|
||||
// Protects |renderer_| in Start(), Stop() and OnClose().
|
||||
rtc::CriticalSection lock_;
|
||||
|
||||
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
|
||||
};
|
||||
|
||||
class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
|
||||
public:
|
||||
explicit WebRtcAudioReceiveStream(int voe_channel_id)
|
||||
: channel_(voe_channel_id) {}
|
||||
|
||||
int channel() { return channel_; }
|
||||
|
||||
private:
|
||||
int channel_;
|
||||
|
||||
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
|
||||
};
|
||||
|
||||
// WebRtcVoiceMediaChannel
|
||||
@ -1417,8 +1445,8 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
|
||||
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
|
||||
|
||||
// Remove any remaining send streams.
|
||||
while (!send_channels_.empty()) {
|
||||
RemoveSendStream(send_channels_.begin()->first);
|
||||
while (!send_streams_.empty()) {
|
||||
RemoveSendStream(send_streams_.begin()->first);
|
||||
}
|
||||
|
||||
// Remove any remaining receive streams.
|
||||
@ -1481,6 +1509,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
|
||||
}
|
||||
}
|
||||
|
||||
// TODO(solenberg): Don't recreate unless options changed.
|
||||
RecreateAudioReceiveStreams();
|
||||
|
||||
LOG(LS_INFO) << "Set voice channel options. Current options: "
|
||||
@ -1760,7 +1789,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
||||
|
||||
// Cache the codecs in order to configure the channel created later.
|
||||
send_codecs_ = codecs;
|
||||
for (const auto& ch : send_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
if (!SetSendCodecs(ch.second->channel(), codecs)) {
|
||||
return false;
|
||||
}
|
||||
@ -1871,7 +1900,7 @@ bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
|
||||
return true;
|
||||
}
|
||||
|
||||
for (const auto& ch : send_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
|
||||
return false;
|
||||
}
|
||||
@ -1935,8 +1964,9 @@ bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
|
||||
desired_send_ = send;
|
||||
if (!send_channels_.empty())
|
||||
if (!send_streams_.empty()) {
|
||||
return ChangeSend(desired_send_);
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
@ -1959,7 +1989,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
|
||||
}
|
||||
|
||||
// Change the settings on each send channel.
|
||||
for (const auto& ch : send_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
if (!ChangeSend(ch.second->channel(), send)) {
|
||||
return false;
|
||||
}
|
||||
@ -2075,13 +2105,13 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
|
||||
return false;
|
||||
}
|
||||
|
||||
// Save the channel to send_channels_, so that RemoveSendStream() can still
|
||||
// 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_channels_.insert(
|
||||
send_streams_.insert(
|
||||
std::make_pair(ssrc,
|
||||
new WebRtcVoiceChannelRenderer(channel, audio_transport)));
|
||||
new WebRtcAudioSendStream(channel, audio_transport, ssrc, 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
|
||||
@ -2094,7 +2124,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
|
||||
// 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_channels_.size() == 1) {
|
||||
if (send_streams_.size() == 1) {
|
||||
receiver_reports_ssrc_ = ssrc;
|
||||
for (const auto& ch : receive_channels_) {
|
||||
int recv_channel = ch.second->channel();
|
||||
@ -2113,8 +2143,8 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::iterator it = send_channels_.find(ssrc);
|
||||
if (it == send_channels_.end()) {
|
||||
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;
|
||||
@ -2126,7 +2156,7 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
|
||||
// Delete the WebRtcVoiceChannelRenderer object connected to the channel,
|
||||
// this will disconnect the audio renderer with the send channel.
|
||||
delete it->second;
|
||||
send_channels_.erase(it);
|
||||
send_streams_.erase(it);
|
||||
|
||||
// Clean up and delete the send channel.
|
||||
LOG(LS_INFO) << "Removing audio send stream " << ssrc
|
||||
@ -2134,7 +2164,7 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
|
||||
if (!DeleteChannel(channel)) {
|
||||
return false;
|
||||
}
|
||||
if (send_channels_.empty()) {
|
||||
if (send_streams_.empty()) {
|
||||
ChangeSend(SEND_NOTHING);
|
||||
}
|
||||
return true;
|
||||
@ -2176,11 +2206,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
|
||||
return false;
|
||||
}
|
||||
|
||||
webrtc::AudioTransport* audio_transport =
|
||||
engine()->voe()->base()->audio_transport();
|
||||
WebRtcVoiceChannelRenderer* channel_renderer =
|
||||
new WebRtcVoiceChannelRenderer(channel, audio_transport);
|
||||
receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
|
||||
WebRtcAudioReceiveStream* stream = new WebRtcAudioReceiveStream(channel);
|
||||
receive_channels_.insert(std::make_pair(ssrc, stream));
|
||||
receive_stream_params_[ssrc] = sp;
|
||||
AddAudioReceiveStream(ssrc);
|
||||
|
||||
@ -2249,7 +2276,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
|
||||
|
||||
ChannelMap::iterator it = receive_channels_.find(ssrc);
|
||||
auto it = receive_channels_.find(ssrc);
|
||||
if (it == receive_channels_.end()) {
|
||||
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
|
||||
<< " which doesn't exist.";
|
||||
@ -2259,9 +2286,6 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
|
||||
RemoveAudioReceiveStream(ssrc);
|
||||
receive_stream_params_.erase(ssrc);
|
||||
|
||||
// Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
|
||||
// will disconnect the audio renderer with the receive channel.
|
||||
// Cache the channel before the deletion.
|
||||
const int channel = it->second->channel();
|
||||
delete it->second;
|
||||
receive_channels_.erase(it);
|
||||
@ -2278,8 +2302,8 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
|
||||
|
||||
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
|
||||
AudioRenderer* renderer) {
|
||||
ChannelMap::iterator it = send_channels_.find(ssrc);
|
||||
if (it == send_channels_.end()) {
|
||||
auto it = send_streams_.find(ssrc);
|
||||
if (it == send_streams_.end()) {
|
||||
if (renderer) {
|
||||
// Return an error if trying to set a valid renderer with an invalid ssrc.
|
||||
LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
|
||||
@ -2388,8 +2412,8 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
|
||||
if (flags & cricket::DF_SEND) {
|
||||
int channel = -1;
|
||||
if (ssrc == 0) {
|
||||
if (send_channels_.size() > 0) {
|
||||
channel = send_channels_.begin()->second->channel();
|
||||
if (send_streams_.size() > 0) {
|
||||
channel = send_streams_.begin()->second->channel();
|
||||
}
|
||||
} else {
|
||||
channel = GetSendChannelId(ssrc);
|
||||
@ -2499,7 +2523,7 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
|
||||
// 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_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
engine()->voe()->network()->ReceivedRTCPPacket(
|
||||
ch.second->channel(), packet->data(), packet->size());
|
||||
}
|
||||
@ -2521,7 +2545,7 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
|
||||
// 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_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
if (!all_muted) {
|
||||
break;
|
||||
}
|
||||
@ -2570,7 +2594,7 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
|
||||
if (is_multi_rate) {
|
||||
// If codec is multi-rate then just set the bitrate.
|
||||
codec.rate = bps;
|
||||
for (const auto& ch : send_channels_) {
|
||||
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.";
|
||||
@ -2630,7 +2654,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
|
||||
webrtc::CodecInst codec;
|
||||
unsigned int level;
|
||||
|
||||
for (const auto& ch : send_channels_) {
|
||||
for (const auto& ch : send_streams_) {
|
||||
const int channel = ch.second->channel();
|
||||
|
||||
// Fill in the sender info, based on what we know, and what the
|
||||
@ -2791,7 +2815,7 @@ int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::const_iterator it = receive_channels_.find(ssrc);
|
||||
const auto it = receive_channels_.find(ssrc);
|
||||
if (it != receive_channels_.end()) {
|
||||
return it->second->channel();
|
||||
}
|
||||
@ -2800,8 +2824,8 @@ int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
|
||||
|
||||
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
ChannelMap::const_iterator it = send_channels_.find(ssrc);
|
||||
if (it != send_channels_.end()) {
|
||||
const auto it = send_streams_.find(ssrc);
|
||||
if (it != send_streams_.end()) {
|
||||
return it->second->channel();
|
||||
}
|
||||
return -1;
|
||||
@ -2927,8 +2951,8 @@ void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() {
|
||||
|
||||
void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
|
||||
RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
||||
WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
|
||||
RTC_DCHECK(channel != nullptr);
|
||||
WebRtcAudioReceiveStream* stream = receive_channels_[ssrc];
|
||||
RTC_DCHECK(stream != nullptr);
|
||||
RTC_DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
|
||||
webrtc::AudioReceiveStream::Config config;
|
||||
config.rtp.remote_ssrc = ssrc;
|
||||
@ -2936,7 +2960,7 @@ void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
|
||||
config.rtp.extensions = recv_rtp_extensions_;
|
||||
config.combined_audio_video_bwe =
|
||||
options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
|
||||
config.voe_channel_id = channel->channel();
|
||||
config.voe_channel_id = stream->channel();
|
||||
config.sync_group = receive_stream_params_[ssrc].sync_label;
|
||||
webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
|
||||
receive_streams_.insert(std::make_pair(ssrc, s));
|
||||
|
||||
@ -172,6 +172,8 @@ class WebRtcVoiceEngine
|
||||
Settable<bool> extended_filter_aec_;
|
||||
Settable<bool> delay_agnostic_aec_;
|
||||
Settable<bool> experimental_ns_;
|
||||
|
||||
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
|
||||
};
|
||||
|
||||
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
|
||||
@ -265,11 +267,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
||||
bool SetPlayout(int channel, bool playout);
|
||||
static Error WebRtcErrorToChannelError(int err_code);
|
||||
|
||||
class WebRtcVoiceChannelRenderer;
|
||||
// Map of ssrc to WebRtcVoiceChannelRenderer object. A new object of
|
||||
// WebRtcVoiceChannelRenderer will be created for every new stream and
|
||||
// will be destroyed when the stream goes away.
|
||||
typedef std::map<uint32_t, WebRtcVoiceChannelRenderer*> ChannelMap;
|
||||
typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
|
||||
unsigned char);
|
||||
|
||||
@ -327,12 +324,12 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
||||
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
|
||||
uint32_t receiver_reports_ssrc_ = 1;
|
||||
|
||||
// send_channels_ contains the channels which are being used for sending.
|
||||
// When the default channel (default_send_channel_id) is used for sending, it
|
||||
// is contained in send_channels_, otherwise not.
|
||||
ChannelMap send_channels_;
|
||||
class WebRtcAudioSendStream;
|
||||
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
|
||||
std::vector<RtpHeaderExtension> send_extensions_;
|
||||
ChannelMap receive_channels_;
|
||||
|
||||
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
|
||||
@ -340,6 +337,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
||||
// Reads on the worker thread are ok.
|
||||
std::vector<RtpHeaderExtension> receive_extensions_;
|
||||
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
|
||||
|
||||
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
|
||||
};
|
||||
|
||||
} // namespace cricket
|
||||
|
||||
@ -59,6 +59,7 @@ static const cricket::AudioCodec* const kAudioCodecs[] = {
|
||||
};
|
||||
static uint32_t kSsrc1 = 0x99;
|
||||
static uint32_t kSsrc2 = 0x98;
|
||||
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
|
||||
|
||||
class FakeVoEWrapper : public cricket::VoEWrapper {
|
||||
public:
|
||||
@ -1862,44 +1863,39 @@ TEST_F(WebRtcVoiceEngineTestFake, Playout) {
|
||||
TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
|
||||
SetupForMultiSendStream();
|
||||
|
||||
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
|
||||
|
||||
// Set the global state for sending.
|
||||
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
|
||||
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
EXPECT_NE(nullptr, call_.GetAudioSendStream(ssrc));
|
||||
|
||||
// Verify that we are in a sending state for all the created streams.
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
|
||||
EXPECT_TRUE(voe_.GetSend(channel_num));
|
||||
}
|
||||
EXPECT_EQ(ARRAY_SIZE(kSsrcs4), call_.GetAudioSendStreams().size());
|
||||
|
||||
// Remove the first send channel, which is the default channel. It will only
|
||||
// recycle the default channel but not delete it.
|
||||
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[0]));
|
||||
// Stream should already be Removed from the send stream list.
|
||||
EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[0]));
|
||||
|
||||
// Delete the rest of send channel streams.
|
||||
for (unsigned int i = 1; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[i]));
|
||||
// Delete the send streams.
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->RemoveSendStream(ssrc));
|
||||
EXPECT_EQ(nullptr, call_.GetAudioSendStream(ssrc));
|
||||
// Stream should already be deleted.
|
||||
EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[i]));
|
||||
EXPECT_EQ(-1, voe_.GetChannelFromLocalSsrc(kSsrcs4[i]));
|
||||
EXPECT_FALSE(channel_->RemoveSendStream(ssrc));
|
||||
EXPECT_EQ(-1, voe_.GetChannelFromLocalSsrc(ssrc));
|
||||
}
|
||||
EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
|
||||
}
|
||||
|
||||
// Test SetSendCodecs correctly configure the codecs in all send streams.
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
||||
SetupForMultiSendStream();
|
||||
|
||||
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
|
||||
// Create send streams.
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
}
|
||||
|
||||
cricket::AudioSendParameters parameters;
|
||||
@ -1911,8 +1907,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
||||
|
||||
// Verify ISAC and VAD are corrected configured on all send channels.
|
||||
webrtc::CodecInst gcodec;
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_STREQ("ISAC", gcodec.plname);
|
||||
EXPECT_TRUE(voe_.GetVAD(channel_num));
|
||||
@ -1922,8 +1918,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
||||
// Change to PCMU(8K) and CN(16K). VAD should not be activated.
|
||||
parameters.codecs[0] = kPcmuCodec;
|
||||
EXPECT_TRUE(channel_->SetSendParameters(parameters));
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
|
||||
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
|
||||
EXPECT_STREQ("PCMU", gcodec.plname);
|
||||
EXPECT_FALSE(voe_.GetVAD(channel_num));
|
||||
@ -1934,28 +1930,27 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
|
||||
TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
|
||||
SetupForMultiSendStream();
|
||||
|
||||
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
|
||||
// Create the send channels and they should be a SEND_NOTHING date.
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
int channel_num = voe_.GetLastChannel();
|
||||
EXPECT_FALSE(voe_.GetSend(channel_num));
|
||||
}
|
||||
|
||||
// Set the global state for starting sending.
|
||||
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
// Verify that we are in a sending state for all the send streams.
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
|
||||
EXPECT_TRUE(voe_.GetSend(channel_num));
|
||||
}
|
||||
|
||||
// Set the global state for stopping sending.
|
||||
EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
|
||||
for (unsigned int i = 1; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
// Verify that we are in a stop state for all the send streams.
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(kSsrcs4[i]);
|
||||
int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
|
||||
EXPECT_FALSE(voe_.GetSend(channel_num));
|
||||
}
|
||||
}
|
||||
@ -1964,11 +1959,10 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
|
||||
TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
|
||||
SetupForMultiSendStream();
|
||||
|
||||
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
|
||||
// Create send streams.
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
for (uint32_t ssrc : kSsrcs4) {
|
||||
EXPECT_TRUE(channel_->AddSendStream(
|
||||
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
|
||||
cricket::StreamParams::CreateLegacy(ssrc)));
|
||||
}
|
||||
// Create a receive stream to check that none of the send streams end up in
|
||||
// the receive stream stats.
|
||||
@ -1983,6 +1977,7 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
|
||||
EXPECT_EQ(static_cast<size_t>(ARRAY_SIZE(kSsrcs4)), info.senders.size());
|
||||
|
||||
// Verify the statistic information is correct.
|
||||
// TODO(solenberg): Make this loop ordering independent.
|
||||
for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs4); ++i) {
|
||||
EXPECT_EQ(kSsrcs4[i], info.senders[i].ssrc());
|
||||
EXPECT_EQ(kPcmuCodec.name, info.senders[i].codec_name);
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user