| /* |
| * Copyright (C) 2017 Apple Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright |
| * notice, this list of conditions and the following disclaimer in the |
| * documentation and/or other materials provided with the distribution. |
| * |
| * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' |
| * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, |
| * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR |
| * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS |
| * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR |
| * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF |
| * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS |
| * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN |
| * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) |
| * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF |
| * THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #include "config.h" |
| #include "LibWebRTCAudioModule.h" |
| |
| #if USE(LIBWEBRTC) |
| |
| #include "LibWebRTCAudioFormat.h" |
| #include "Logging.h" |
| #include <wtf/FastMalloc.h> |
| |
| #if PLATFORM(COCOA) |
| #include "IncomingAudioMediaStreamTrackRendererUnit.h" |
| #endif |
| |
| namespace WebCore { |
| |
| LibWebRTCAudioModule::LibWebRTCAudioModule() |
| : m_queue(WorkQueue::create("WebKitWebRTCAudioModule", WorkQueue::QOS::UserInteractive)) |
| , m_logTimer(*this, &LibWebRTCAudioModule::logTimerFired) |
| { |
| } |
| |
| LibWebRTCAudioModule::~LibWebRTCAudioModule() |
| { |
| } |
| |
| int32_t LibWebRTCAudioModule::RegisterAudioCallback(webrtc::AudioTransport* audioTransport) |
| { |
| RELEASE_LOG(WebRTC, "LibWebRTCAudioModule::RegisterAudioCallback %d", !!audioTransport); |
| |
| m_audioTransport = audioTransport; |
| return 0; |
| } |
| |
| int32_t LibWebRTCAudioModule::StartPlayout() |
| { |
| RELEASE_LOG(WebRTC, "LibWebRTCAudioModule::StartPlayout %d", m_isPlaying); |
| |
| if (m_isPlaying) |
| return 0; |
| |
| m_isPlaying = true; |
| callOnMainThread([this, protectedThis = rtc::scoped_refptr<webrtc::AudioDeviceModule>(this)] { |
| m_logTimer.startRepeating(logTimerInterval); |
| }); |
| |
| m_queue->dispatch([this, protectedThis = rtc::scoped_refptr<webrtc::AudioDeviceModule>(this)] { |
| m_pollingTime = MonotonicTime::now(); |
| #if PLATFORM(COCOA) |
| m_currentAudioSampleCount = 0; |
| #endif |
| pollAudioData(); |
| }); |
| return 0; |
| } |
| |
| int32_t LibWebRTCAudioModule::StopPlayout() |
| { |
| RELEASE_LOG(WebRTC, "LibWebRTCAudioModule::StopPlayout %d", m_isPlaying); |
| |
| m_isPlaying = false; |
| callOnMainThread([this, protectedThis = rtc::scoped_refptr<webrtc::AudioDeviceModule>(this)] { |
| m_logTimer.stop(); |
| }); |
| return 0; |
| } |
| |
| void LibWebRTCAudioModule::logTimerFired() |
| { |
| RELEASE_LOG_IF(m_timeSpent, WebRTC, "LibWebRTCAudioModule::pollAudioData, polling took too much time: %d ms", m_timeSpent); |
| m_timeSpent = 0; |
| } |
| |
| // libwebrtc uses 10ms frames. |
| const unsigned frameLengthMs = 1000 * LibWebRTCAudioFormat::chunkSampleCount / LibWebRTCAudioFormat::sampleRate; |
| const unsigned pollInterval = LibWebRTCAudioModule::PollSamplesCount * frameLengthMs; |
| const unsigned channels = 2; |
| |
| Seconds LibWebRTCAudioModule::computeDelayUntilNextPolling() |
| { |
| auto now = MonotonicTime::now(); |
| auto delayUntilNextPolling = m_pollingTime + Seconds::fromMilliseconds(pollInterval) - now; |
| if (delayUntilNextPolling.milliseconds() < 0) { |
| m_timeSpent = (now - m_pollingTime).milliseconds(); |
| delayUntilNextPolling = 0_s; |
| } |
| m_pollingTime = now + delayUntilNextPolling; |
| return delayUntilNextPolling; |
| } |
| |
| void LibWebRTCAudioModule::pollAudioData() |
| { |
| if (!m_isPlaying) |
| return; |
| |
| Function<void()> nextPollFunction = [this, protectedThis = rtc::scoped_refptr<webrtc::AudioDeviceModule>(this)] { |
| pollAudioData(); |
| }; |
| |
| { |
| // For performance reasons, we forbid heap allocations while doing rendering on the webrtc audio thread. |
| ForbidMallocUseForCurrentThreadScope forbidMallocUse; |
| |
| pollFromSource(); |
| } |
| m_queue->dispatchAfter(computeDelayUntilNextPolling(), WTFMove(nextPollFunction)); |
| } |
| |
| void LibWebRTCAudioModule::pollFromSource() |
| { |
| if (!m_audioTransport) |
| return; |
| |
| for (unsigned i = 0; i < PollSamplesCount; i++) { |
| int64_t elapsedTime = -1; |
| int64_t ntpTime = -1; |
| char data[LibWebRTCAudioFormat::sampleByteSize * channels * LibWebRTCAudioFormat::chunkSampleCount]; |
| m_audioTransport->PullRenderData(LibWebRTCAudioFormat::sampleByteSize * 8, LibWebRTCAudioFormat::sampleRate, channels, LibWebRTCAudioFormat::chunkSampleCount, data, &elapsedTime, &ntpTime); |
| #if PLATFORM(COCOA) |
| if (m_isRenderingIncomingAudio) |
| m_incomingAudioMediaStreamTrackRendererUnit->newAudioChunkPushed(m_currentAudioSampleCount); |
| m_currentAudioSampleCount += LibWebRTCAudioFormat::chunkSampleCount; |
| #endif |
| } |
| } |
| |
| #if PLATFORM(COCOA) |
| BaseAudioMediaStreamTrackRendererUnit& LibWebRTCAudioModule::incomingAudioMediaStreamTrackRendererUnit() |
| { |
| if (!m_incomingAudioMediaStreamTrackRendererUnit) |
| m_incomingAudioMediaStreamTrackRendererUnit = makeUnique<IncomingAudioMediaStreamTrackRendererUnit>(*this); |
| return *m_incomingAudioMediaStreamTrackRendererUnit; |
| } |
| #endif |
| |
| } // namespace WebCore |
| |
| #endif // USE(LIBWEBRTC) |