- 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}
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index a0386b0..6e2a88f 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -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() {
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 @@
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::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(
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index fb271f2..de56a03 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -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 @@
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 @@
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_;
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5375358..eed1195 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1313,15 +1313,25 @@
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 @@
// 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 @@
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 @@
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 @@
// 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 @@
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 @@
}
}
+ // TODO(solenberg): Don't recreate unless options changed.
RecreateAudioReceiveStreams();
LOG(LS_INFO) << "Set voice channel options. Current options: "
@@ -1760,7 +1789,7 @@
// 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 @@
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::SetSend(SendFlags send) {
desired_send_ = send;
- if (!send_channels_.empty())
+ if (!send_streams_.empty()) {
return ChangeSend(desired_send_);
+ }
return true;
}
@@ -1959,7 +1989,7 @@
}
// 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 @@
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 @@
// 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::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 @@
// 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 @@
if (!DeleteChannel(channel)) {
return false;
}
- if (send_channels_.empty()) {
+ if (send_streams_.empty()) {
ChangeSend(SEND_NOTHING);
}
return true;
@@ -2176,11 +2206,8 @@
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 @@
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 @@
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::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 @@
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 @@
// 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 @@
// 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 @@
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 @@
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::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::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::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 @@
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));
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index a8f3ec8..a708405 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -172,6 +172,8 @@
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 @@
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 @@
// 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 @@
// 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
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 85094f8..477369e 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -59,6 +59,7 @@
};
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, 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 @@
// 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 @@
// 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, 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, 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 @@
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);
