commit d5247510dc9cf1b1f8eb1f79310bef702b7804c1
author Fredrik Solenberg <solenberg@webrtc.org> Mon Dec 18 21:41:03 2017
committer Commit Bot <commit-bot@chromium.org> Mon Dec 18 22:51:27 2017
tree 89c22b2efafa1774333f0c03a46bf6997a3d8c28
parent 086c9f5e4e6b8a77927f80a8e692fa440f940adf

Replace VoEBase::[Start/Stop]Playout().

The functionality is moved into AudioState.

TBR: henrika@webrtc.org
Bug: webrtc:4690
Change-Id: I015482ad18a39609634f6ead9e991d5210107f0f
Reviewed-on: https://webrtc-review.googlesource.com/34502
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21338}

audio/audio_state.cc

diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index ac7eaf7..9d980e8 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -14,6 +14,7 @@
 #include <utility>
 #include <vector>
 
+#include "audio/audio_receive_stream.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/atomicops.h"
 #include "rtc_base/checks.h"
@@ -37,6 +38,7 @@
 
 AudioState::~AudioState() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(receiving_streams_.empty());
   RTC_DCHECK(sending_streams_.empty());
 }
 
@@ -45,16 +47,44 @@
   return config_.voice_engine;
 }
 
-rtc::scoped_refptr<AudioMixer> AudioState::mixer() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  return config_.audio_mixer;
-}
-
 bool AudioState::typing_noise_detected() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return audio_transport_.typing_noise_detected();
 }
 
+void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
+  receiving_streams_.insert(stream);
+  if (!config_.audio_mixer->AddSource(
+      static_cast<internal::AudioReceiveStream*>(stream))) {
+    RTC_LOG(LS_ERROR) << "Failed to add source to mixer.";
+  }
+
+  // Make sure playback is initialized; start playing if enabled.
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Playing()) {
+    if (adm->InitPlayout() == 0) {
+      if (playout_enabled_) {
+        adm->StartPlayout();
+      }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize playout.";
+    }
+  }
+}
+
+void AudioState::RemoveReceivingStream(webrtc::AudioReceiveStream* stream) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  auto count = receiving_streams_.erase(stream);
+  RTC_DCHECK_EQ(1, count);
+  config_.audio_mixer->RemoveSource(
+      static_cast<internal::AudioReceiveStream*>(stream));
+  if (receiving_streams_.empty()) {
+    config_.audio_device_module->StopPlayout();
+  }
+}
+
 void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
                                   int sample_rate_hz, size_t num_channels) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ -89,20 +119,18 @@
 void AudioState::SetPlayout(bool enabled) {
   RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  const bool currently_enabled = (null_audio_poller_ == nullptr);
-  if (enabled == currently_enabled) {
-    return;
-  }
-  if (enabled) {
-    null_audio_poller_.reset();
-  }
-  // Will stop/start playout of the underlying device, if necessary, and
-  // remember the setting for when it receives subsequent calls of
-  // StartPlayout.
-  voe_base_->SetPlayout(enabled);
-  if (!enabled) {
-    null_audio_poller_ =
-        rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
+  if (playout_enabled_ != enabled) {
+    playout_enabled_ = enabled;
+    if (enabled) {
+      null_audio_poller_.reset();
+      if (!receiving_streams_.empty()) {
+        config_.audio_device_module->StartPlayout();
+      }
+    } else {
+      config_.audio_device_module->StopPlayout();
+      null_audio_poller_ =
+          rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
+    }
   }
 }
 
@@ -157,7 +185,7 @@
 
 void AudioState::UpdateAudioTransportWithSendingStreams() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  std::vector<AudioSendStream*> sending_streams;
+  std::vector<webrtc::AudioSendStream*> sending_streams;
   int max_sample_rate_hz = 8000;
   size_t max_num_channels = 1;
   for (const auto& kv : sending_streams_) {