This reverts commit c4b391a257.
Reason for revert: issue fixed
Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
This reverts commit 6330818ec8.
Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
Original change's description:
> NetEQ RTP Play: Optionally write output audio file
>
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
>
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
>
> This CL also includes a test script with which the retro-compatibility
> has been verified.
>
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
This CL makes the output audio file optional to more
quickly run neteq_rtpplay when no audio output is needed.
The CL also includes necessary adaptations because of pre-existing
dependencies (e.g., the output audio file name is used to create
the plotting script file names).
The command line arguments are retro-compatible - i.e., same behavior
when specifying the output audio file and the new flag
--output_files_base_name is not used.
This CL also includes a test script with which the retro-compatibility
has been verified.
Bug: webrtc:10337
Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27067}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
I will deprecate deps in proto_library for improved build throughput.
We can use link_deps here instead.
Bug: chromium:938011
Change-Id: Iafa83000c3f7f9ffdc0c376a2297b4a9380b7594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/master@{#26989}
This is a reland of d9f798a6b3
Original change's description:
> Remove field trial include from decision logic.
>
> Bug: webrtc:9289
> Change-Id: I2e465bf9eddda8bde50daeb14cfd51405e536ff4
> Reviewed-on: https://webrtc-review.googlesource.com/c/125097
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26925}
Bug: webrtc:9289
Change-Id: I40fbd999fc8495beaeb46799c333f91d72b5be37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125720
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26978}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
Previously, if more than one packet is extracted in a GetAudio call then
an incorrect number of samples will be reported.
Bug: webrtc:10363
Change-Id: Ia1bcc87a0e0082060e4f746d37a4008735eec6b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124829
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26903}
This replaces the use of command-line flags with the use of a config
struct. This makes it easier for non command-line applications to use
the NetEqTestFactory to run simulations.
Bug: webrtc:10337
Change-Id: I24533bf206e70e12db9af8d9675769c1ff7c7d48
Reviewed-on: https://webrtc-review.googlesource.com/c/123600
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26887}
- This mode estimates relative packet arrival delay for each incoming packet and adds that value to the histogram.
- The histogram buckets are 20 milliseconds each instead of whole packets.
- The functionality is enabled with a field trial for experimentation.
Bug: webrtc:10333
Change-Id: I8f7499c56802fc1aa1ced2f5310fdd2ef1403515
Reviewed-on: https://webrtc-review.googlesource.com/c/123923
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26871}
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.
Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)
The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.
Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
For e.g. when audio receiver is recreated during SetRtpExtensionsAndRecreateStream in webrtc_voice_engine.h,
the audio minimum delay can't go down.
Imagine we set base minimum playout delay when audio receiver stream is created, then its value will be cached, to be applied during recreation. Then SetRtpExtensionsAndRecreateStream is fired, and audio receiver stream is recreated with the cached value, but currently it in the constructor it is used to initialize both base minimum playout delay and minimum playout delay. Which leads to the bug that effective minimum playout delay can't go down anymore as if you set base minimum playout delay to the low value then effective delay use the biggest value which minimum playout delay.
This didn't come up during previous trials because of
https://webrtc-review.googlesource.com/c/src/+/122280
It was reseting minimum playout delay to 0 asynchronously, that is why you couldn't see this bug.
Bug: webrtc:10287
Change-Id: I924446bfcb33ac94f7e5bf987a1868acaf1b0346
Reviewed-on: https://webrtc-review.googlesource.com/c/124000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26832}
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
This reverts commit 5341aaccdb.
Reason for revert: Order of initialization of global static strings.
Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
>
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
>
> Original CL description:
>
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}
The difference to the original is new bitexactness strings AND
global static file string constants. The reason for reland is breaking
downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123387
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26774}
This reverts commit 9c31ac2323.
Reason for revert: Breaks downstream project.
Original change's description:
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
> Reviewed-on: https://webrtc-review.googlesource.com/c/114883
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26742}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I0ac48b7320d31d3e7af33bf8714c3db6c807b82e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123385
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26747}
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
Reviewed-on: https://webrtc-review.googlesource.com/c/114883
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26742}
On NetEq level latency corresponds to delay and two terms can be used interchangeably here.
In order to implement latency constraint we need to provide a range of possible values which should be constant. See getCapabilities() here:
https://www.w3.org/TR/mediacapture-streams/#dfn-applyconstraints-algorithm
Lowest possible value accepted value is constant and equals 0. But because |packet_len_ms_| and |maximum_delay_ms_| may be updated during live of DelayManager upper bound is not constant. Moreover, due to change in |packet_len_ms_| the |minimum_delay_ms_| which was valid when its was set may be considered invalid later on.
To circumvent that and provide constant range for capabilities we separate base minimum delay and minimum delay. ApplyConstraints algorithm will set base minimum delay. Base minimum delay will act as recommendation for lower bound of minimum delay and will be used to limit target delay. If user sets base minimum delay through ApplyConstraints which is bigger than currently
possible maximum (e.g. bigger than NetEq maximum buffer size in milliseconds) then base minimum delay will be clamped to currently possible maximum to match user's intentions as best as possible.
Note, that we keep original behavior when minimum_delay_ms_ (effective_minimum_delay_ms after this CL) in LimitTargetLevel method may be above upper bound due to changing packet audio length.
Bug: webrtc:10287
Change-Id: I06b8f5cd3fd1bc36800af0447f91f7c4dc21a766
Reviewed-on: https://webrtc-review.googlesource.com/c/121700
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26666}
Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.
This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.
There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.
Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26572}
This is first step to allow to set latency
from client code in Chromium.
Existing minimum latency hasn't been used because it can clash
with video syncronization code.
Bug: webrtc:10287
Change-Id: Ia38906506069a1abfa01698dc62df283fc15cfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/121423
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26536}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/111750.
This time we don't use the multistream decoder unless we have to.
(Which is when #channels >2). Pros: don't make downstream projects
crash due to used up stack space, a few % more efficiency for the
typical case (because multistream adds some overhead). Cons: Messy
C-code with "union" types and #define MACROs, probably more
maintenance.
Bug: webrtc:8649
Change-Id: I4253a5e0c382f67ac7c6731dc6602a31e6779e63
Reviewed-on: https://webrtc-review.googlesource.com/c/120049
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26445}
This CL applies clang-tidy's performance-inefficient-vector-operation
[1] on the WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-inefficient-vector-operation.html
Bug: webrtc:10252
Change-Id: I824caab2a5746036852e00d714b89aa5ec030ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/120052
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26442}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This is a reland of 80b95de765
Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
>
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}
Bug: webrtc:6463
Change-Id: I12154ef65744c1b7811974a1d871e05ed3fbbc27
Reviewed-on: https://webrtc-review.googlesource.com/c/118660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26337}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This reverts commit 83ed89a45f.
Reason for revert: breaks downstream project
Original change's description:
> Opus multistream.
>
> This is a backwards-compatible change. It makes WebRTC use the Opus
> multistream decoder for all Opus packets. Single-stream packets are a
> special case of multistream ones (with stream=1).
>
> The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
> 'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
> do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
> did when we had single-stream encoders. Now there may be several
> independent encoders with possibly different BANDWIDTH. The new
> GetMaxPlaybackRate queries all of them, and returns a playback rate if
> all the encoder's rates are equal.
>
> WebRtcOpus_GetSurroundParameters is a configuration convention. It
> maps the number of channels to a multi-stream encoder/decoder
> configuration. As described in RFC 7845
> https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
> encoder/decoder needs a number of streams, number of coupled streams
> and a 255-byte mapping array. The function GetSurroundParameters
> computes all of these from the number of channels. [1, 2, 4, 6, 8]
> channels are supported.
>
> Bug: webrtc:8649
> Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/111750
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26293}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}
