Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index 1147d45..09b0545 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -28,7 +28,6 @@
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/rate_control_settings.h"
-#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
#include "third_party/libyuv/include/libyuv/scale.h"
@@ -444,54 +443,41 @@
return WEBRTC_VIDEO_CODEC_OK;
}
-void SimulcastEncoderAdapter::SetRates(
- const RateControlParameters& parameters) {
+int SimulcastEncoderAdapter::SetRateAllocation(
+ const VideoBitrateAllocation& bitrate,
+ uint32_t new_framerate) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
if (!Initialized()) {
- RTC_LOG(LS_WARNING) << "SetRates while not initialized";
- return;
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
- if (parameters.framerate_fps < 1.0) {
- RTC_LOG(LS_WARNING) << "Invalid framerate: " << parameters.framerate_fps;
- return;
+ if (new_framerate < 1) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
- if (codec_.maxBitrate > 0 &&
- parameters.bitrate.get_sum_kbps() > codec_.maxBitrate) {
- RTC_LOG(LS_WARNING) << "Total bitrate " << parameters.bitrate.get_sum_kbps()
- << " exceeds max bitrate: " << codec_.maxBitrate;
- return;
+ if (codec_.maxBitrate > 0 && bitrate.get_sum_kbps() > codec_.maxBitrate) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
- if (parameters.bitrate.get_sum_bps() > 0) {
+ if (bitrate.get_sum_bps() > 0) {
// Make sure the bitrate fits the configured min bitrates. 0 is a special
// value that means paused, though, so leave it alone.
- if (parameters.bitrate.get_sum_kbps() < codec_.minBitrate) {
- RTC_LOG(LS_WARNING) << "Total bitrate "
- << parameters.bitrate.get_sum_kbps()
- << " is lower than minimum bitrate: "
- << codec_.minBitrate;
- return;
+ if (bitrate.get_sum_kbps() < codec_.minBitrate) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
if (codec_.numberOfSimulcastStreams > 0 &&
- parameters.bitrate.get_sum_kbps() <
- codec_.simulcastStream[0].minBitrate) {
- RTC_LOG(LS_WARNING) << "Total bitrate "
- << parameters.bitrate.get_sum_kbps()
- << " is lower than minimum bitrate of base layer: "
- << codec_.simulcastStream[0].minBitrate;
- return;
+ bitrate.get_sum_kbps() < codec_.simulcastStream[0].minBitrate) {
+ return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
}
}
- codec_.maxFramerate = static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+ codec_.maxFramerate = new_framerate;
for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
uint32_t stream_bitrate_kbps =
- parameters.bitrate.GetSpatialLayerSum(stream_idx) / 1000;
+ bitrate.GetSpatialLayerSum(stream_idx) / 1000;
// Need a key frame if we have not sent this stream before.
if (stream_bitrate_kbps > 0 && !streaminfos_[stream_idx].send_stream) {
@@ -501,31 +487,17 @@
// Slice the temporal layers out of the full allocation and pass it on to
// the encoder handling the current simulcast stream.
- RateControlParameters stream_parameters = parameters;
- stream_parameters.bitrate = VideoBitrateAllocation();
+ VideoBitrateAllocation stream_allocation;
for (int i = 0; i < kMaxTemporalStreams; ++i) {
- if (parameters.bitrate.HasBitrate(stream_idx, i)) {
- stream_parameters.bitrate.SetBitrate(
- 0, i, parameters.bitrate.GetBitrate(stream_idx, i));
+ if (bitrate.HasBitrate(stream_idx, i)) {
+ stream_allocation.SetBitrate(0, i, bitrate.GetBitrate(stream_idx, i));
}
}
-
- // Assign link allocation proportionally to spatial layer allocation.
- if (parameters.bandwidth_allocation != DataRate::Zero()) {
- stream_parameters.bandwidth_allocation =
- DataRate::bps((parameters.bandwidth_allocation.bps() *
- stream_parameters.bitrate.get_sum_bps()) /
- parameters.bitrate.get_sum_bps());
- // Make sure we don't allocate bandwidth lower than target bitrate.
- if (stream_parameters.bandwidth_allocation.bps() <
- stream_parameters.bitrate.get_sum_bps()) {
- stream_parameters.bandwidth_allocation =
- DataRate::bps(stream_parameters.bitrate.get_sum_bps());
- }
- }
-
- streaminfos_[stream_idx].encoder->SetRates(stream_parameters);
+ streaminfos_[stream_idx].encoder->SetRateAllocation(stream_allocation,
+ new_framerate);
}
+
+ return WEBRTC_VIDEO_CODEC_OK;
}
// TODO(brandtr): Add task checker to this member function, when all encoder