Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}diff --git a/audio/null_audio_poller.cc b/audio/null_audio_poller.cc
new file mode 100644
index 0000000..c22b3d8
--- /dev/null
+++ b/audio/null_audio_poller.cc
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/null_audio_poller.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace internal {
+
+namespace {
+
+constexpr int64_t kPollDelayMs = 10; // WebRTC uses 10ms by default
+
+constexpr size_t kNumChannels = 1;
+constexpr uint32_t kSamplesPerSecond = 48000; // 48kHz
+constexpr size_t kNumSamples = kSamplesPerSecond / 100; // 10ms of samples
+
+} // namespace
+
+NullAudioPoller::NullAudioPoller(AudioTransport* audio_transport)
+ : audio_transport_(audio_transport),
+ reschedule_at_(rtc::TimeMillis() + kPollDelayMs) {
+ RTC_DCHECK(audio_transport);
+ OnMessage(nullptr); // Start the poll loop.
+}
+
+NullAudioPoller::~NullAudioPoller() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::Thread::Current()->Clear(this);
+}
+
+void NullAudioPoller::OnMessage(rtc::Message* msg) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
+ // Buffer to hold the audio samples.
+ int16_t buffer[kNumSamples * kNumChannels];
+ // Output variables from |NeedMorePlayData|.
+ size_t n_samples;
+ int64_t elapsed_time_ms;
+ int64_t ntp_time_ms;
+ audio_transport_->NeedMorePlayData(kNumSamples, sizeof(int16_t), kNumChannels,
+ kSamplesPerSecond, buffer, n_samples,
+ &elapsed_time_ms, &ntp_time_ms);
+
+ // Reschedule the next poll iteration. If, for some reason, the given
+ // reschedule time has already passed, reschedule as soon as possible.
+ int64_t now = rtc::TimeMillis();
+ if (reschedule_at_ < now) {
+ reschedule_at_ = now;
+ }
+ rtc::Thread::Current()->PostAt(RTC_FROM_HERE, reschedule_at_, this, 0);
+
+ // Loop after next will be kPollDelayMs later.
+ reschedule_at_ += kPollDelayMs;
+}
+
+} // namespace internal
+} // namespace webrtc