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Revert "Ensure that we always set values for min and max audio bitrate."

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This reverts commit e47aee3b864fe5a4f964d405a7f6f3ac8c49f174.

Reason for revert: Breaks downstream project

Original change's description:
> Ensure that we always set values for min and max audio bitrate.
> 
> Use (in order from lowest to highest precedence):
> -- fixed 32000bps
> -- fixed target bitrate from codec
> -- explicit values from the rtp encoding parameters
> -- Final precedence is given to field trial values from
>    WebRTC-Audio-Allocation
> 
> Bug: webrtc:10487
> Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
> Reviewed-by: Minyue Li <minyue@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Daniel Lee <dklee@google.com>
> Cr-Commit-Position: refs/heads/master@{#27667}

TBR=solenberg@webrtc.org,stefan@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,minyue@webrtc.org,minyue@google.com,dklee@google.com

Change-Id: Ie975cf40e65105d1e4cfab417b220b6bfc34592b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27670}
This commit is contained in:
Daniel Lee
2019-04-17 15:27:05 +00:00
committed by Commit Bot
parent b4b53bd8cc
commit 63658d06ec
7 changed files with 120 additions and 194 deletions
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36
rtc_base/experiments/audio_allocation_settings.cc
View File

@ -12,6 +12,9 @@
namespace webrtc {
namespace {
// For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000.
const int kOpusMinBitrateBps = 6000;
const int kOpusBitrateFbBps = 32000;
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
} // namespace
@ -20,8 +23,8 @@ AudioAllocationSettings::AudioAllocationSettings()
allocate_audio_without_feedback_("Enabled"),
force_no_audio_feedback_("Enabled"),
send_side_bwe_with_overhead_("Enabled"),
min_bitrate_("min"),
max_bitrate_("max"),
default_min_bitrate_("min", DataRate::bps(kOpusMinBitrateBps)),
default_max_bitrate_("max", DataRate::bps(kOpusBitrateFbBps)),
priority_bitrate_("prio", DataRate::Zero()) {
ParseFieldTrial({&audio_send_side_bwe_},
field_trial::FindFullName("WebRTC-Audio-SendSideBwe"));
@ -32,8 +35,9 @@ AudioAllocationSettings::AudioAllocationSettings()
ParseFieldTrial({&send_side_bwe_with_overhead_},
field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
ParseFieldTrial({&min_bitrate_, &max_bitrate_, &priority_bitrate_},
field_trial::FindFullName("WebRTC-Audio-Allocation"));
ParseFieldTrial(
{&default_min_bitrate_, &default_max_bitrate_, &priority_bitrate_},
field_trial::FindFullName("WebRTC-Audio-Allocation"));
// TODO(mflodman): Keep testing this and set proper values.
// Note: This is an early experiment currently only supported by Opus.
@ -96,15 +100,25 @@ bool AudioAllocationSettings::IncludeAudioInAllocationOnReconfigure(
return true;
}
bool AudioAllocationSettings::IncludeOverheadInAudioAllocation() const {
return send_side_bwe_with_overhead_;
int AudioAllocationSettings::MinBitrateBps() const {
return default_min_bitrate_->bps() + min_overhead_bps_;
}
absl::optional<DataRate> AudioAllocationSettings::MinBitrate() const {
return min_bitrate_.GetOptional();
}
absl::optional<DataRate> AudioAllocationSettings::MaxBitrate() const {
return max_bitrate_.GetOptional();
int AudioAllocationSettings::MaxBitrateBps(
absl::optional<int> rtp_parameter_max_bitrate_bps) const {
// We assume that the max is a hard limit on the payload bitrate, so we add
// min_overhead_bps to it to ensure that, when overhead is deducted, the
// payload rate never goes beyond the limit. Note: this also means that if a
// higher overhead is forced, we cannot reach the limit.
// TODO(minyue): Reconsider this when the signaling to BWE is done
// through a dedicated API.
// This means that when RtpParameters is reset, we may change the
// encoder's bit rate immediately (through ReconfigureAudioSendStream()),
// meanwhile change the cap to the output of BWE.
if (rtp_parameter_max_bitrate_bps)
return *rtp_parameter_max_bitrate_bps + min_overhead_bps_;
return default_max_bitrate_->bps() + min_overhead_bps_;
}
DataRate AudioAllocationSettings::DefaultPriorityBitrate() const {
DataRate max_overhead = DataRate::Zero();

21
rtc_base/experiments/audio_allocation_settings.h
View File

@ -60,13 +60,14 @@ class AudioAllocationSettings {
int max_bitrate_bps,
bool has_dscp,
int transport_seq_num_extension_header_id) const;
// Returns true if we should include packet overhead in audio allocation.
bool IncludeOverheadInAudioAllocation() const;
// Returns the min bitrate for audio rate allocation.
absl::optional<DataRate> MinBitrate() const;
// Returns the max bitrate for audio rate allocation.
absl::optional<DataRate> MaxBitrate() const;
// Returns the min bitrate for audio rate allocation, potentially including
// overhead.
int MinBitrateBps() const;
// Returns the max bitrate for audio rate allocation, potentially including
// overhead. |rtp_parameter_max_bitrate_bps| max bitrate as configured in rtp
// parameters, excluding overhead.
int MaxBitrateBps(absl::optional<int> rtp_parameter_max_bitrate_bps) const;
// Indicates the default priority bitrate for audio streams. The bitrate
// allocator will prioritize audio until it reaches this bitrate and will
// divide bitrate evently between audio and video above this bitrate.
@ -78,10 +79,10 @@ class AudioAllocationSettings {
FieldTrialFlag force_no_audio_feedback_;
FieldTrialFlag send_side_bwe_with_overhead_;
int min_overhead_bps_ = 0;
// Field Trial configured bitrates to use as overrides over default/user
// configured bitrate range when audio bitrate allocation is enabled.
FieldTrialOptional<DataRate> min_bitrate_;
FieldTrialOptional<DataRate> max_bitrate_;
// Default bitrates to use as range if there's no user configured
// bitrate range but audio bitrate allocation is enabled.
FieldTrialParameter<DataRate> default_min_bitrate_;
FieldTrialParameter<DataRate> default_max_bitrate_;
FieldTrialParameter<DataRate> priority_bitrate_;
};
} // namespace webrtc