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Remove priority_rate from AudioStreamConfig.

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This API is going away, we'll use the WebRTC-Audio-Allocation field
trial flag to set this value in the future.

Bug: webrtc:10556
Change-Id: I2c4c1948a33f909fac069dd038cea36a793e4745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145405
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28608}
This commit is contained in:
Jonas Olsson 2019-07-18 13:09:49 +02:00 committed by Commit Bot
parent 824fb38b9f
commit 857ad62721
4 changed files with 0 additions and 13 deletions
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3
test/scenario/audio_stream.cc
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@ -127,9 +127,6 @@ SendAudioStream::SendAudioStream(
{RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});
}
if (config.encoder.priority_rate) {
send_config.track_id = sender->GetNextPriorityId();
}
sender_->SendTask([&] {
send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {

7
test/scenario/call_client.cc
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@ -29,8 +29,6 @@ const uint32_t kVideoRecvLocalSsrcs[kNumSsrcs] = {0xDAB001, 0xDAB002, 0xDAB003,
const uint32_t kAudioSendSsrc = 0xDEADBEEF;
const uint32_t kReceiverLocalAudioSsrc = 0x1234567;
const char* kPriorityStreamId = "priority-track";
constexpr int kEventLogOutputIntervalMs = 5000;
CallClientFakeAudio InitAudio(TimeController* time_controller) {
@ -317,11 +315,6 @@ uint32_t CallClient::GetNextRtxSsrc() {
return kSendRtxSsrcs[next_rtx_ssrc_index_++];
}
std::string CallClient::GetNextPriorityId() {
RTC_CHECK_LT(next_priority_index_++, 1);
return kPriorityStreamId;
}
void CallClient::AddExtensions(std::vector<RtpExtension> extensions) {
for (const auto& extension : extensions)
header_parser_->RegisterRtpHeaderExtension(extension);

2
test/scenario/call_client.h
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@ -128,7 +128,6 @@ class CallClient : public EmulatedNetworkReceiverInterface {
uint32_t GetNextAudioSsrc();
uint32_t GetNextAudioLocalSsrc();
uint32_t GetNextRtxSsrc();
std::string GetNextPriorityId();
void AddExtensions(std::vector<RtpExtension> extensions);
void SendTask(std::function<void()> task);
@ -150,7 +149,6 @@ class CallClient : public EmulatedNetworkReceiverInterface {
int next_rtx_ssrc_index_ = 0;
int next_audio_ssrc_index_ = 0;
int next_audio_local_ssrc_index_ = 0;
int next_priority_index_ = 0;
std::map<uint32_t, MediaType> ssrc_media_types_;
// Defined last so it's destroyed first.
TaskQueueForTest task_queue_;

1
test/scenario/scenario_config.h
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@ -202,7 +202,6 @@ struct AudioStreamConfig {
absl::optional<DataRate> fixed_rate;
absl::optional<DataRate> min_rate;
absl::optional<DataRate> max_rate;
absl::optional<DataRate> priority_rate;
TimeDelta initial_frame_length = TimeDelta::ms(20);
} encoder;
struct Stream {