зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1253499 - Break out VideoStreamFactory to separate file. r=dminor
Differential Revision: https://phabricator.services.mozilla.com/D6263 --HG-- extra : moz-landing-system : lando
This commit is contained in:
Andreas Pehrson
Родитель
6f26c10cda
Коммит
ac369ba83b
|
|
@ -8,6 +8,7 @@
|
|||
|
||||
#include "AudioConduit.h"
|
||||
#include "VideoConduit.h"
|
||||
#include "VideoStreamFactory.h"
|
||||
#include "YuvStamper.h"
|
||||
#include "mozilla/TemplateLib.h"
|
||||
#include "mozilla/media/MediaUtils.h"
|
||||
|
|
@ -88,12 +89,8 @@ static const char* kRedPayloadName = "red";
|
|||
// is active and one or more layers are being scaled.
|
||||
#define SIMULCAST_RESOLUTION_ALIGNMENT 16
|
||||
|
||||
// Convert (SI) kilobits/sec to (SI) bits/sec
|
||||
#define KBPS(kbps) kbps * 1000
|
||||
|
||||
// 32 bytes is what WebRTC CodecInst expects
|
||||
const unsigned int WebrtcVideoConduit::CODEC_PLNAME_SIZE = 32;
|
||||
static const int kViEMinCodecBitrate_bps = KBPS(30);
|
||||
|
||||
template<typename T>
|
||||
T MinIgnoreZero(const T& a, const T& b)
|
||||
|
|
@ -173,100 +170,6 @@ SelectSendFrameRate(const VideoCodecConfig* codecConfig,
|
|||
return new_framerate;
|
||||
}
|
||||
|
||||
#define MB_OF(w,h) ((unsigned int)((((w+15)>>4))*((unsigned int)((h+15)>>4))))
|
||||
// For now, try to set the max rates well above the knee in the curve.
|
||||
// Chosen somewhat arbitrarily; it's hard to find good data oriented for
|
||||
// realtime interactive/talking-head recording. These rates assume
|
||||
// 30fps.
|
||||
|
||||
// XXX Populate this based on a pref (which we should consider sorting because
|
||||
// people won't assume they need to).
|
||||
static WebrtcVideoConduit::ResolutionAndBitrateLimits kResolutionAndBitrateLimits[] = {
|
||||
{MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
|
||||
{MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
|
||||
{MB_OF(800, 480), KBPS(600), KBPS(800), KBPS(2500)}, // HD ~720
|
||||
{MB_OF(480, 270), KBPS(150), KBPS(500), KBPS(2000)}, // WVGA
|
||||
{tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(125), KBPS(300), KBPS(1300)}, // VGA
|
||||
{MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
|
||||
{0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
|
||||
};
|
||||
|
||||
static WebrtcVideoConduit::ResolutionAndBitrateLimits
|
||||
GetLimitsFor(unsigned int aWidth, unsigned int aHeight, int aCapBps = 0)
|
||||
{
|
||||
// max bandwidth should be proportional (not linearly!) to resolution, and
|
||||
// proportional (perhaps linearly, or close) to current frame rate.
|
||||
int fs = MB_OF(aWidth, aHeight);
|
||||
|
||||
for (const auto& resAndLimits : kResolutionAndBitrateLimits) {
|
||||
if (fs > resAndLimits.resolution_in_mb &&
|
||||
// pick the highest range where at least start rate is within cap
|
||||
// (or if we're at the end of the array).
|
||||
(aCapBps == 0 ||
|
||||
resAndLimits.start_bitrate_bps <= aCapBps ||
|
||||
resAndLimits.resolution_in_mb == 0)) {
|
||||
return resAndLimits;
|
||||
}
|
||||
}
|
||||
|
||||
MOZ_CRASH("Loop should have handled fallback");
|
||||
}
|
||||
|
||||
/**
|
||||
* Function to set the encoding bitrate limits based on incoming frame size and rate
|
||||
* @param width, height: dimensions of the frame
|
||||
* @param min: minimum bitrate in bps
|
||||
* @param start: bitrate in bps that the encoder should start with
|
||||
* @param cap: user-enforced max bitrate, or 0
|
||||
* @param pref_cap: cap enforced by prefs
|
||||
* @param negotiated_cap: cap negotiated through SDP
|
||||
* @param aVideoStream stream to apply bitrates to
|
||||
*/
|
||||
static void
|
||||
SelectBitrates(
|
||||
unsigned short width, unsigned short height,
|
||||
int min, int start,
|
||||
int cap, int pref_cap, int negotiated_cap,
|
||||
webrtc::VideoStream& aVideoStream)
|
||||
{
|
||||
int& out_min = aVideoStream.min_bitrate_bps;
|
||||
int& out_start = aVideoStream.target_bitrate_bps;
|
||||
int& out_max = aVideoStream.max_bitrate_bps;
|
||||
|
||||
WebrtcVideoConduit::ResolutionAndBitrateLimits resAndLimits =
|
||||
GetLimitsFor(width, height);
|
||||
out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
|
||||
out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
|
||||
out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
|
||||
|
||||
// Note: negotiated_cap is the max transport bitrate - it applies to
|
||||
// a single codec encoding, but should also apply to the sum of all
|
||||
// simulcast layers in this encoding! So sum(layers.maxBitrate) <=
|
||||
// negotiated_cap
|
||||
// Note that out_max already has had pref_cap applied to it
|
||||
out_max = MinIgnoreZero(negotiated_cap, out_max);
|
||||
out_min = std::min(out_min, out_max);
|
||||
out_start = std::min(out_start, out_max);
|
||||
|
||||
if (min && min > out_min) {
|
||||
out_min = min;
|
||||
}
|
||||
// If we try to set a minimum bitrate that is too low, ViE will reject it.
|
||||
out_min = std::max(kViEMinCodecBitrate_bps, out_min);
|
||||
out_max = std::max(kViEMinCodecBitrate_bps, out_max);
|
||||
if (start && start > out_start) {
|
||||
out_start = start;
|
||||
}
|
||||
|
||||
// Ensure that min <= start <= max
|
||||
if (out_min > out_max) {
|
||||
out_min = out_max;
|
||||
}
|
||||
out_start = std::min(out_max, std::max(out_start, out_min));
|
||||
|
||||
MOZ_ASSERT(pref_cap == 0 || out_max <= pref_cap);
|
||||
}
|
||||
|
||||
/**
|
||||
* Perform validation on the codecConfig to be applied
|
||||
*/
|
||||
|
|
@ -898,135 +801,6 @@ CodecsDifferent(const nsTArray<UniquePtr<VideoCodecConfig>>& a,
|
|||
return false;
|
||||
}
|
||||
|
||||
std::vector<webrtc::VideoStream>
|
||||
WebrtcVideoConduit::VideoStreamFactory::CreateEncoderStreams(
|
||||
int width, int height, const webrtc::VideoEncoderConfig& config)
|
||||
{
|
||||
size_t streamCount = config.number_of_streams;
|
||||
|
||||
// We only allow one layer when screensharing
|
||||
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
|
||||
streamCount = 1;
|
||||
}
|
||||
|
||||
std::vector<webrtc::VideoStream> streams;
|
||||
streams.reserve(streamCount);
|
||||
|
||||
// We assume that the first stream is the full-resolution stream.
|
||||
|
||||
// This ensures all simulcast layers will be of the same aspect ratio as the input.
|
||||
mSimulcastAdapter->OnOutputFormatRequest(
|
||||
cricket::VideoFormat(width, height, 0, 0));
|
||||
|
||||
for (size_t idx = streamCount - 1; streamCount > 0; idx--, streamCount--) {
|
||||
webrtc::VideoStream video_stream;
|
||||
auto& simulcastEncoding = mCodecConfig.mSimulcastEncodings[idx];
|
||||
MOZ_ASSERT(simulcastEncoding.constraints.scaleDownBy >= 1.0);
|
||||
|
||||
// All streams' dimensions must retain the aspect ratio of the input stream.
|
||||
// Note that the first stream might already have been scaled by us.
|
||||
// Webrtc.org doesn't know this, so we have to adjust lower layers manually.
|
||||
int unusedCropWidth, unusedCropHeight, outWidth, outHeight;
|
||||
if (idx == 0) {
|
||||
// This is the highest-resolution stream. We avoid calling
|
||||
// AdaptFrameResolution on this because precision errors in VideoAdapter
|
||||
// can cause the out-resolution to be an odd pixel smaller than the
|
||||
// source (1920x1419 has caused this). We shortcut this instead.
|
||||
outWidth = width;
|
||||
outHeight = height;
|
||||
} else {
|
||||
float effectiveScaleDownBy =
|
||||
simulcastEncoding.constraints.scaleDownBy /
|
||||
mCodecConfig.mSimulcastEncodings[0].constraints.scaleDownBy;
|
||||
MOZ_ASSERT(effectiveScaleDownBy >= 1.0);
|
||||
mSimulcastAdapter->OnScaleResolutionBy(
|
||||
effectiveScaleDownBy > 1.0 ?
|
||||
rtc::Optional<float>(effectiveScaleDownBy) :
|
||||
rtc::Optional<float>());
|
||||
bool rv = mSimulcastAdapter->AdaptFrameResolution(
|
||||
width,
|
||||
height,
|
||||
0, // Ok, since we don't request an output format with an interval
|
||||
&unusedCropWidth,
|
||||
&unusedCropHeight,
|
||||
&outWidth,
|
||||
&outHeight);
|
||||
|
||||
if (!rv) {
|
||||
// The only thing that can make AdaptFrameResolution fail in this case
|
||||
// is if this layer is scaled so far down that it has less than one pixel.
|
||||
outWidth = 0;
|
||||
outHeight = 0;
|
||||
}
|
||||
}
|
||||
|
||||
if (outWidth == 0 || outHeight == 0) {
|
||||
CSFLogInfo(LOGTAG,
|
||||
"%s Stream with RID %s ignored because of no resolution.",
|
||||
__FUNCTION__, simulcastEncoding.rid.c_str());
|
||||
continue;
|
||||
}
|
||||
|
||||
MOZ_ASSERT(outWidth > 0);
|
||||
MOZ_ASSERT(outHeight > 0);
|
||||
video_stream.width = outWidth;
|
||||
video_stream.height = outHeight;
|
||||
|
||||
CSFLogInfo(LOGTAG, "%s Input frame %ux%u, RID %s scaling to %zux%zu",
|
||||
__FUNCTION__, width, height, simulcastEncoding.rid.c_str(),
|
||||
video_stream.width, video_stream.height);
|
||||
|
||||
if (video_stream.width * height != width * video_stream.height) {
|
||||
CSFLogInfo(LOGTAG,
|
||||
"%s Stream with RID %s ignored because of bad aspect ratio.",
|
||||
__FUNCTION__, simulcastEncoding.rid.c_str());
|
||||
continue;
|
||||
}
|
||||
|
||||
// We want to ensure this picks up the current framerate, so indirect
|
||||
video_stream.max_framerate = mSendingFramerate;
|
||||
|
||||
SelectBitrates(
|
||||
video_stream.width, video_stream.height,
|
||||
mMinBitrate, mStartBitrate, simulcastEncoding.constraints.maxBr,
|
||||
mPrefMaxBitrate, mNegotiatedMaxBitrate, video_stream);
|
||||
|
||||
video_stream.max_qp = kQpMax;
|
||||
video_stream.SetRid(simulcastEncoding.rid);
|
||||
|
||||
// leave vector temporal_layer_thresholds_bps empty for non-simulcast
|
||||
video_stream.temporal_layer_thresholds_bps.clear();
|
||||
if (config.number_of_streams > 1) {
|
||||
// XXX Note: in simulcast.cc in upstream code, the array value is
|
||||
// 3(-1) for all streams, though it's in an array, except for screencasts,
|
||||
// which use 1 (i.e 2 layers).
|
||||
|
||||
// Oddly, though this is a 'bps' array, nothing really looks at the
|
||||
// values for normal video, just the size of the array to know the
|
||||
// number of temporal layers.
|
||||
// For VideoEncoderConfig::ContentType::kScreen, though, in
|
||||
// video_codec_initializer.cc it uses [0] to set the target bitrate
|
||||
// for the screenshare.
|
||||
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
|
||||
video_stream.temporal_layer_thresholds_bps.push_back(video_stream.target_bitrate_bps);
|
||||
} else {
|
||||
video_stream.temporal_layer_thresholds_bps.resize(2);
|
||||
}
|
||||
// XXX Bug 1390215 investigate using more of
|
||||
// simulcast.cc:GetSimulcastConfig() or our own algorithm to replace it
|
||||
}
|
||||
|
||||
if (mCodecConfig.mName == "H264") {
|
||||
if (mCodecConfig.mEncodingConstraints.maxMbps > 0) {
|
||||
// Not supported yet!
|
||||
CSFLogError(LOGTAG, "%s H.264 max_mbps not supported yet", __FUNCTION__);
|
||||
}
|
||||
}
|
||||
streams.push_back(video_stream);
|
||||
}
|
||||
return streams;
|
||||
}
|
||||
|
||||
/**
|
||||
* Note: Setting the send-codec on the Video Engine will restart the encoder,
|
||||
* sets up new SSRC and reset RTP_RTCP module with the new codec setting.
|
||||
|
|
@ -2654,7 +2428,7 @@ WebrtcVideoConduit::VideoEncoderConfigBuilder::SetEncoderSpecificSettings(
|
|||
}
|
||||
|
||||
void
|
||||
WebrtcVideoConduit::VideoEncoderConfigBuilder::SetVideoStreamFactory(rtc::scoped_refptr<WebrtcVideoConduit::VideoStreamFactory> aFactory)
|
||||
WebrtcVideoConduit::VideoEncoderConfigBuilder::SetVideoStreamFactory(rtc::scoped_refptr<VideoStreamFactory> aFactory)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
|
||||
|
|
|
|||
|
|
@ -40,9 +40,17 @@
|
|||
|
||||
namespace mozilla {
|
||||
|
||||
// Convert (SI) kilobits/sec to (SI) bits/sec
|
||||
#define KBPS(kbps) kbps * 1000
|
||||
|
||||
const int kViEMinCodecBitrate_bps = KBPS(30);
|
||||
const unsigned int kVideoMtu = 1200;
|
||||
const int kQpMax = 56;
|
||||
|
||||
template<typename T>
|
||||
T MinIgnoreZero(const T& a, const T& b);
|
||||
|
||||
class VideoStreamFactory;
|
||||
class WebrtcAudioConduit;
|
||||
class nsThread;
|
||||
|
||||
|
|
@ -68,14 +76,6 @@ class WebrtcVideoConduit : public VideoSessionConduit
|
|||
, public rtc::VideoSourceInterface<webrtc::VideoFrame>
|
||||
{
|
||||
public:
|
||||
struct ResolutionAndBitrateLimits
|
||||
{
|
||||
int resolution_in_mb;
|
||||
int min_bitrate_bps;
|
||||
int start_bitrate_bps;
|
||||
int max_bitrate_bps;
|
||||
};
|
||||
|
||||
//VoiceEngine defined constant for Payload Name Size.
|
||||
static const unsigned int CODEC_PLNAME_SIZE;
|
||||
|
||||
|
|
@ -421,8 +421,6 @@ private:
|
|||
* Stores encoder configuration information and produces
|
||||
* a VideoEncoderConfig from it.
|
||||
*/
|
||||
class VideoStreamFactory;
|
||||
|
||||
class VideoEncoderConfigBuilder {
|
||||
public:
|
||||
/**
|
||||
|
|
@ -434,7 +432,7 @@ private:
|
|||
double jsScaleDownBy=1.0; // user-controlled downscale
|
||||
};
|
||||
void SetEncoderSpecificSettings(rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings> aSettings);
|
||||
void SetVideoStreamFactory(rtc::scoped_refptr<WebrtcVideoConduit::VideoStreamFactory> aFactory);
|
||||
void SetVideoStreamFactory(rtc::scoped_refptr<VideoStreamFactory> aFactory);
|
||||
void SetMinTransmitBitrateBps(int aXmitMinBps);
|
||||
void SetContentType(webrtc::VideoEncoderConfig::ContentType aContentType);
|
||||
void SetResolutionDivisor(unsigned char aDivisor);
|
||||
|
|
@ -456,65 +454,6 @@ private:
|
|||
// Utility function to dump recv codec database
|
||||
void DumpCodecDB() const;
|
||||
|
||||
// Factory class for VideoStreams... vie_encoder.cc will call this to reconfigure.
|
||||
class VideoStreamFactory : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface
|
||||
{
|
||||
public:
|
||||
VideoStreamFactory(VideoCodecConfig aConfig,
|
||||
webrtc::VideoCodecMode aCodecMode,
|
||||
int aMinBitrate, int aStartBitrate,
|
||||
int aPrefMaxBitrate, int aNegotiatedMaxBitrate,
|
||||
unsigned int aSendingFramerate)
|
||||
: mCodecMode(aCodecMode)
|
||||
, mSendingFramerate(aSendingFramerate)
|
||||
, mCodecConfig(std::forward<VideoCodecConfig>(aConfig))
|
||||
, mMinBitrate(aMinBitrate)
|
||||
, mStartBitrate(aStartBitrate)
|
||||
, mPrefMaxBitrate(aPrefMaxBitrate)
|
||||
, mNegotiatedMaxBitrate(aNegotiatedMaxBitrate)
|
||||
, mSimulcastAdapter(MakeUnique<cricket::VideoAdapter>())
|
||||
{}
|
||||
|
||||
void SetCodecMode(webrtc::VideoCodecMode aCodecMode)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
mCodecMode = aCodecMode;
|
||||
}
|
||||
|
||||
void SetSendingFramerate(unsigned int aSendingFramerate)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
mSendingFramerate = aSendingFramerate;
|
||||
}
|
||||
|
||||
private:
|
||||
// This gets called off-main thread and may hold internal webrtc.org
|
||||
// locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
|
||||
std::vector<webrtc::VideoStream>
|
||||
CreateEncoderStreams(int width, int height,
|
||||
const webrtc::VideoEncoderConfig& config) override;
|
||||
|
||||
// Used to limit number of streams for screensharing.
|
||||
Atomic<webrtc::VideoCodecMode> mCodecMode;
|
||||
|
||||
// The framerate we're currently sending at.
|
||||
Atomic<unsigned int> mSendingFramerate;
|
||||
|
||||
// The current send codec config, containing simulcast layer configs.
|
||||
const VideoCodecConfig mCodecConfig;
|
||||
|
||||
// Bitrate limits in bps.
|
||||
const int mMinBitrate = 0;
|
||||
const int mStartBitrate = 0;
|
||||
const int mPrefMaxBitrate = 0;
|
||||
const int mNegotiatedMaxBitrate = 0;
|
||||
|
||||
// Adapter for simulcast layers. We use this to handle scaleResolutionDownBy
|
||||
// for layers. It's separate from the conduit's mVideoAdapter to not affect
|
||||
// scaling settings for incoming frames.
|
||||
UniquePtr<cricket::VideoAdapter> mSimulcastAdapter;
|
||||
};
|
||||
|
||||
// Video Latency Test averaging filter
|
||||
void VideoLatencyUpdate(uint64_t new_sample);
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,258 @@
|
|||
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
|
||||
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
|
||||
/* This Source Code Form is subject to the terms of the Mozilla Public
|
||||
* License, v. 2.0. If a copy of the MPL was not distributed with this
|
||||
* file, You can obtain one at https://mozilla.org/MPL/2.0/. */
|
||||
|
||||
#include "VideoStreamFactory.h"
|
||||
|
||||
#include "CSFLog.h"
|
||||
#include "nsThreadUtils.h"
|
||||
#include "VideoConduit.h"
|
||||
|
||||
namespace mozilla {
|
||||
|
||||
#ifdef LOGTAG
|
||||
#undef LOGTAG
|
||||
#endif
|
||||
#define LOGTAG "WebrtcVideoSessionConduit"
|
||||
|
||||
#define MB_OF(w,h) ((unsigned int)((((w+15)>>4))*((unsigned int)((h+15)>>4))))
|
||||
// For now, try to set the max rates well above the knee in the curve.
|
||||
// Chosen somewhat arbitrarily; it's hard to find good data oriented for
|
||||
// realtime interactive/talking-head recording. These rates assume
|
||||
// 30fps.
|
||||
|
||||
// XXX Populate this based on a pref (which we should consider sorting because
|
||||
// people won't assume they need to).
|
||||
static VideoStreamFactory::ResolutionAndBitrateLimits kResolutionAndBitrateLimits[] = {
|
||||
{MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
|
||||
{MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
|
||||
{MB_OF(800, 480), KBPS(600), KBPS(800), KBPS(2500)}, // HD ~720
|
||||
{MB_OF(480, 270), KBPS(150), KBPS(500), KBPS(2000)}, // WVGA
|
||||
{tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(125), KBPS(300), KBPS(1300)}, // VGA
|
||||
{MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
|
||||
{0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
|
||||
};
|
||||
|
||||
static VideoStreamFactory::ResolutionAndBitrateLimits
|
||||
GetLimitsFor(unsigned int aWidth, unsigned int aHeight, int aCapBps = 0)
|
||||
{
|
||||
// max bandwidth should be proportional (not linearly!) to resolution, and
|
||||
// proportional (perhaps linearly, or close) to current frame rate.
|
||||
int fs = MB_OF(aWidth, aHeight);
|
||||
|
||||
for (const auto& resAndLimits : kResolutionAndBitrateLimits) {
|
||||
if (fs > resAndLimits.resolution_in_mb &&
|
||||
// pick the highest range where at least start rate is within cap
|
||||
// (or if we're at the end of the array).
|
||||
(aCapBps == 0 ||
|
||||
resAndLimits.start_bitrate_bps <= aCapBps ||
|
||||
resAndLimits.resolution_in_mb == 0)) {
|
||||
return resAndLimits;
|
||||
}
|
||||
}
|
||||
|
||||
MOZ_CRASH("Loop should have handled fallback");
|
||||
}
|
||||
|
||||
/**
|
||||
* Function to set the encoding bitrate limits based on incoming frame size and rate
|
||||
* @param width, height: dimensions of the frame
|
||||
* @param min: minimum bitrate in bps
|
||||
* @param start: bitrate in bps that the encoder should start with
|
||||
* @param cap: user-enforced max bitrate, or 0
|
||||
* @param pref_cap: cap enforced by prefs
|
||||
* @param negotiated_cap: cap negotiated through SDP
|
||||
* @param aVideoStream stream to apply bitrates to
|
||||
*/
|
||||
static void
|
||||
SelectBitrates(
|
||||
unsigned short width, unsigned short height,
|
||||
int min, int start,
|
||||
int cap, int pref_cap, int negotiated_cap,
|
||||
webrtc::VideoStream& aVideoStream)
|
||||
{
|
||||
int& out_min = aVideoStream.min_bitrate_bps;
|
||||
int& out_start = aVideoStream.target_bitrate_bps;
|
||||
int& out_max = aVideoStream.max_bitrate_bps;
|
||||
|
||||
VideoStreamFactory::ResolutionAndBitrateLimits resAndLimits =
|
||||
GetLimitsFor(width, height);
|
||||
out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
|
||||
out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
|
||||
out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
|
||||
|
||||
// Note: negotiated_cap is the max transport bitrate - it applies to
|
||||
// a single codec encoding, but should also apply to the sum of all
|
||||
// simulcast layers in this encoding! So sum(layers.maxBitrate) <=
|
||||
// negotiated_cap
|
||||
// Note that out_max already has had pref_cap applied to it
|
||||
out_max = MinIgnoreZero(negotiated_cap, out_max);
|
||||
out_min = std::min(out_min, out_max);
|
||||
out_start = std::min(out_start, out_max);
|
||||
|
||||
if (min && min > out_min) {
|
||||
out_min = min;
|
||||
}
|
||||
// If we try to set a minimum bitrate that is too low, ViE will reject it.
|
||||
out_min = std::max(kViEMinCodecBitrate_bps, out_min);
|
||||
out_max = std::max(kViEMinCodecBitrate_bps, out_max);
|
||||
if (start && start > out_start) {
|
||||
out_start = start;
|
||||
}
|
||||
|
||||
// Ensure that min <= start <= max
|
||||
if (out_min > out_max) {
|
||||
out_min = out_max;
|
||||
}
|
||||
out_start = std::min(out_max, std::max(out_start, out_min));
|
||||
|
||||
MOZ_ASSERT(pref_cap == 0 || out_max <= pref_cap);
|
||||
}
|
||||
|
||||
void
|
||||
VideoStreamFactory::SetCodecMode(webrtc::VideoCodecMode aCodecMode)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
mCodecMode = aCodecMode;
|
||||
}
|
||||
|
||||
void
|
||||
VideoStreamFactory::SetSendingFramerate(unsigned int aSendingFramerate)
|
||||
{
|
||||
MOZ_ASSERT(NS_IsMainThread());
|
||||
mSendingFramerate = aSendingFramerate;
|
||||
}
|
||||
|
||||
std::vector<webrtc::VideoStream>
|
||||
VideoStreamFactory::CreateEncoderStreams(
|
||||
int width, int height, const webrtc::VideoEncoderConfig& config)
|
||||
{
|
||||
size_t streamCount = config.number_of_streams;
|
||||
|
||||
// We only allow one layer when screensharing
|
||||
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
|
||||
streamCount = 1;
|
||||
}
|
||||
|
||||
std::vector<webrtc::VideoStream> streams;
|
||||
streams.reserve(streamCount);
|
||||
|
||||
// We assume that the first stream is the full-resolution stream.
|
||||
|
||||
// This ensures all simulcast layers will be of the same aspect ratio as the input.
|
||||
mSimulcastAdapter->OnOutputFormatRequest(
|
||||
cricket::VideoFormat(width, height, 0, 0));
|
||||
|
||||
for (size_t idx = streamCount - 1; streamCount > 0; idx--, streamCount--) {
|
||||
webrtc::VideoStream video_stream;
|
||||
auto& simulcastEncoding = mCodecConfig.mSimulcastEncodings[idx];
|
||||
MOZ_ASSERT(simulcastEncoding.constraints.scaleDownBy >= 1.0);
|
||||
|
||||
// All streams' dimensions must retain the aspect ratio of the input stream.
|
||||
// Note that the first stream might already have been scaled by us.
|
||||
// Webrtc.org doesn't know this, so we have to adjust lower layers manually.
|
||||
int unusedCropWidth, unusedCropHeight, outWidth, outHeight;
|
||||
if (idx == 0) {
|
||||
// This is the highest-resolution stream. We avoid calling
|
||||
// AdaptFrameResolution on this because precision errors in VideoAdapter
|
||||
// can cause the out-resolution to be an odd pixel smaller than the
|
||||
// source (1920x1419 has caused this). We shortcut this instead.
|
||||
outWidth = width;
|
||||
outHeight = height;
|
||||
} else {
|
||||
float effectiveScaleDownBy =
|
||||
simulcastEncoding.constraints.scaleDownBy /
|
||||
mCodecConfig.mSimulcastEncodings[0].constraints.scaleDownBy;
|
||||
MOZ_ASSERT(effectiveScaleDownBy >= 1.0);
|
||||
mSimulcastAdapter->OnScaleResolutionBy(
|
||||
effectiveScaleDownBy > 1.0 ?
|
||||
rtc::Optional<float>(effectiveScaleDownBy) :
|
||||
rtc::Optional<float>());
|
||||
bool rv = mSimulcastAdapter->AdaptFrameResolution(
|
||||
width,
|
||||
height,
|
||||
0, // Ok, since we don't request an output format with an interval
|
||||
&unusedCropWidth,
|
||||
&unusedCropHeight,
|
||||
&outWidth,
|
||||
&outHeight);
|
||||
|
||||
if (!rv) {
|
||||
// The only thing that can make AdaptFrameResolution fail in this case
|
||||
// is if this layer is scaled so far down that it has less than one pixel.
|
||||
outWidth = 0;
|
||||
outHeight = 0;
|
||||
}
|
||||
}
|
||||
|
||||
if (outWidth == 0 || outHeight == 0) {
|
||||
CSFLogInfo(LOGTAG,
|
||||
"%s Stream with RID %s ignored because of no resolution.",
|
||||
__FUNCTION__, simulcastEncoding.rid.c_str());
|
||||
continue;
|
||||
}
|
||||
|
||||
MOZ_ASSERT(outWidth > 0);
|
||||
MOZ_ASSERT(outHeight > 0);
|
||||
video_stream.width = outWidth;
|
||||
video_stream.height = outHeight;
|
||||
|
||||
CSFLogInfo(LOGTAG, "%s Input frame %ux%u, RID %s scaling to %zux%zu",
|
||||
__FUNCTION__, width, height, simulcastEncoding.rid.c_str(),
|
||||
video_stream.width, video_stream.height);
|
||||
|
||||
if (video_stream.width * height != width * video_stream.height) {
|
||||
CSFLogInfo(LOGTAG,
|
||||
"%s Stream with RID %s ignored because of bad aspect ratio.",
|
||||
__FUNCTION__, simulcastEncoding.rid.c_str());
|
||||
continue;
|
||||
}
|
||||
|
||||
// We want to ensure this picks up the current framerate, so indirect
|
||||
video_stream.max_framerate = mSendingFramerate;
|
||||
|
||||
SelectBitrates(
|
||||
video_stream.width, video_stream.height,
|
||||
mMinBitrate, mStartBitrate, simulcastEncoding.constraints.maxBr,
|
||||
mPrefMaxBitrate, mNegotiatedMaxBitrate, video_stream);
|
||||
|
||||
video_stream.max_qp = kQpMax;
|
||||
video_stream.SetRid(simulcastEncoding.rid);
|
||||
|
||||
// leave vector temporal_layer_thresholds_bps empty for non-simulcast
|
||||
video_stream.temporal_layer_thresholds_bps.clear();
|
||||
if (config.number_of_streams > 1) {
|
||||
// XXX Note: in simulcast.cc in upstream code, the array value is
|
||||
// 3(-1) for all streams, though it's in an array, except for screencasts,
|
||||
// which use 1 (i.e 2 layers).
|
||||
|
||||
// Oddly, though this is a 'bps' array, nothing really looks at the
|
||||
// values for normal video, just the size of the array to know the
|
||||
// number of temporal layers.
|
||||
// For VideoEncoderConfig::ContentType::kScreen, though, in
|
||||
// video_codec_initializer.cc it uses [0] to set the target bitrate
|
||||
// for the screenshare.
|
||||
if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
|
||||
video_stream.temporal_layer_thresholds_bps.push_back(video_stream.target_bitrate_bps);
|
||||
} else {
|
||||
video_stream.temporal_layer_thresholds_bps.resize(2);
|
||||
}
|
||||
// XXX Bug 1390215 investigate using more of
|
||||
// simulcast.cc:GetSimulcastConfig() or our own algorithm to replace it
|
||||
}
|
||||
|
||||
if (mCodecConfig.mName == "H264") {
|
||||
if (mCodecConfig.mEncodingConstraints.maxMbps > 0) {
|
||||
// Not supported yet!
|
||||
CSFLogError(LOGTAG, "%s H.264 max_mbps not supported yet", __FUNCTION__);
|
||||
}
|
||||
}
|
||||
streams.push_back(video_stream);
|
||||
}
|
||||
return streams;
|
||||
}
|
||||
|
||||
} // namespace mozilla
|
||||
|
||||
|
|
@ -0,0 +1,79 @@
|
|||
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
|
||||
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
|
||||
/* This Source Code Form is subject to the terms of the Mozilla Public
|
||||
* License, v. 2.0. If a copy of the MPL was not distributed with this
|
||||
* file, You can obtain one at https://mozilla.org/MPL/2.0/. */
|
||||
|
||||
#ifndef VideoStreamFactory_h
|
||||
#define VideoStreamFactory_h
|
||||
|
||||
#include "CodecConfig.h"
|
||||
#include "mozilla/Atomics.h"
|
||||
#include "mozilla/UniquePtr.h"
|
||||
#include "webrtc/config.h"
|
||||
#include "webrtc/media/base/videoadapter.h"
|
||||
|
||||
namespace mozilla {
|
||||
|
||||
// Factory class for VideoStreams... vie_encoder.cc will call this to reconfigure.
|
||||
class VideoStreamFactory : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface
|
||||
{
|
||||
public:
|
||||
struct ResolutionAndBitrateLimits
|
||||
{
|
||||
int resolution_in_mb;
|
||||
int min_bitrate_bps;
|
||||
int start_bitrate_bps;
|
||||
int max_bitrate_bps;
|
||||
};
|
||||
|
||||
VideoStreamFactory(VideoCodecConfig aConfig,
|
||||
webrtc::VideoCodecMode aCodecMode,
|
||||
int aMinBitrate, int aStartBitrate,
|
||||
int aPrefMaxBitrate, int aNegotiatedMaxBitrate,
|
||||
unsigned int aSendingFramerate)
|
||||
: mCodecMode(aCodecMode)
|
||||
, mSendingFramerate(aSendingFramerate)
|
||||
, mCodecConfig(std::forward<VideoCodecConfig>(aConfig))
|
||||
, mMinBitrate(aMinBitrate)
|
||||
, mStartBitrate(aStartBitrate)
|
||||
, mPrefMaxBitrate(aPrefMaxBitrate)
|
||||
, mNegotiatedMaxBitrate(aNegotiatedMaxBitrate)
|
||||
, mSimulcastAdapter(MakeUnique<cricket::VideoAdapter>())
|
||||
{}
|
||||
|
||||
void SetCodecMode(webrtc::VideoCodecMode aCodecMode);
|
||||
void SetSendingFramerate(unsigned int aSendingFramerate);
|
||||
|
||||
// This gets called off-main thread and may hold internal webrtc.org
|
||||
// locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
|
||||
std::vector<webrtc::VideoStream>
|
||||
CreateEncoderStreams(int width, int height,
|
||||
const webrtc::VideoEncoderConfig& config) override;
|
||||
|
||||
private:
|
||||
// Used to limit number of streams for screensharing.
|
||||
Atomic<webrtc::VideoCodecMode> mCodecMode;
|
||||
|
||||
// The framerate we're currently sending at.
|
||||
Atomic<unsigned int> mSendingFramerate;
|
||||
|
||||
// The current send codec config, containing simulcast layer configs.
|
||||
const VideoCodecConfig mCodecConfig;
|
||||
|
||||
// Bitrate limits in bps.
|
||||
const int mMinBitrate = 0;
|
||||
const int mStartBitrate = 0;
|
||||
const int mPrefMaxBitrate = 0;
|
||||
const int mNegotiatedMaxBitrate = 0;
|
||||
|
||||
// Adapter for simulcast layers. We use this to handle scaleResolutionDownBy
|
||||
// for layers. It's separate from the conduit's mVideoAdapter to not affect
|
||||
// scaling settings for incoming frames.
|
||||
UniquePtr<cricket::VideoAdapter> mSimulcastAdapter;
|
||||
};
|
||||
|
||||
} // namespace mozilla
|
||||
|
||||
#endif
|
||||
|
||||
|
|
@ -24,6 +24,7 @@ UNIFIED_SOURCES += [
|
|||
'MediaDataDecoderCodec.cpp',
|
||||
'RtpSourceObserver.cpp',
|
||||
'VideoConduit.cpp',
|
||||
'VideoStreamFactory.cpp',
|
||||
'WebrtcGmpVideoCodec.cpp',
|
||||
'WebrtcMediaDataDecoderCodec.cpp',
|
||||
]
|
||||
|
|
|
|||
Загрузка…
Ссылка в новой задаче