We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1349393003
Cr-Commit-Position: refs/heads/master@{#10046}
A couple of places were missing handling of UYVY video formats.
BUG=webrtc:4816
Review URL: https://codereview.webrtc.org/1317613003
Cr-Commit-Position: refs/heads/master@{#10044}
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
* Make sure they're all final and don't allow copying or assignment.
* Get rid of the single-channel PCM decoder classes.
* Move some includes from .h to .cc files where possible.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1353803002
Cr-Commit-Position: refs/heads/master@{#10021}
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1339253003 .
Cr-Commit-Position: refs/heads/master@{#10018}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348613003
Cr-Commit-Position: refs/heads/master@{#10015}
Video capture for android is now implemented in talk/app/webrtc/androidvideocapturer.h
BUG=webrtc:4475
Review URL: https://codereview.webrtc.org/1347083003
Cr-Commit-Position: refs/heads/master@{#9995}
The callback object was not used anymore. Also removing the deprecated
WEBRTC_DTMF_DETECTION macro from engine_configurations.h.
BUG=3520
Review URL: https://codereview.webrtc.org/1353763002
Cr-Commit-Position: refs/heads/master@{#9988}
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.
In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.
BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1327933003 .
Cr-Commit-Position: refs/heads/master@{#9984}
The functions were essentially no-op. Also removing forward declaration
of ACMDTMFDetection, which was not used.
BUG=3520
Review URL: https://codereview.webrtc.org/1356543003
Cr-Commit-Position: refs/heads/master@{#9982}
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
that we can open up audio in communication mode also on older
devices that only supports it in combination with 16kHz.
BUG=webrtc:4756
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1347243003 .
Cr-Commit-Position: refs/heads/master@{#9971}
The unit test currently works as follows:
RtxLoopBackTransport logs the sequence numbers for all sent packets in expected_sequence_numbers_. Since the transport is configured to drop some of the packets there will be requests for retransmissions and the RTX sequence numbers will also be stored in the same list.
The (non-rtx) packets are received by VerifyingRtxReceiver which also stores the sequence numbers in a list sequence_numbers_. Both lists are then sorted and sequence_numbers_ is compared to whatever is in the start of expected_sequence_numbers_.
This works assuming that the RTX sequence numbers are greater than the regular RTP sequence numbers. In the RTP sender, both RTP and RTX are set to start at "random" 15-bit sequence numbers. The RTP sequence number is then changed to 2345 in the unit test, which would imply that the RTX sequence number is lower than the ones for RTP with probability ~1%. The reason why the test works anyway is that the test sets up a fake clock, which is used to initialize the random number generator in RTPSender, and the fixed starting point for the clock happens to result in RTX sequence numbers greater than 2345. However, any change to the initialization of the sequence numbers, the seeding of the PRNG or the fake clock causes a test failure with probability ~1%.
The new code omits the RTX sequence numbers from expected_sequence_numbers_, thus avoiding the problem with low RTX sequence numbers. The initialization of the sequence numbers in RTPSender is also bad, but I'll fix that in another CL.
Review URL: https://codereview.webrtc.org/1263383002
Cr-Commit-Position: refs/heads/master@{#9967}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1346993002
Cr-Commit-Position: refs/heads/master@{#9966}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348053002
Cr-Commit-Position: refs/heads/master@{#9961}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
Remove start_bitrate_bps which is no longer used and return the current
allocated bitrate instead of having it as an out parameter, removing the
previous return value which is no longer used.
Permits removing bitrate controller usage from ViEEncoder.
BUG=webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1343783006 .
Cr-Commit-Position: refs/heads/master@{#9942}
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
Helps differentiate between different instances when debugging.
Review URL: https://codereview.webrtc.org/1337003003
Cr-Commit-Position: refs/heads/master@{#9927}
Collects packet information within a struct instead of spreading it out
over different vectors. Adds a fixed-size buffer to the stored packet
instead of using vectors.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1340573002
Cr-Commit-Position: refs/heads/master@{#9926}
This CL ensures that we return -1 in cases where InitRecording() fails. It ensures that we don't crash applications.
BUG=b/22849644
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1323243012 .
Cr-Commit-Position: refs/heads/master@{#9918}
If we don't, we'll end up crashing if they're enabled when the speech
encoder is in the middle of encoding a packet, since CNG and RED
assume that the speech encoder starts out with an empty buffer
(because they need to be in sync with it).
BUG=chromium:490368
Review URL: https://codereview.webrtc.org/1331853002
Cr-Commit-Position: refs/heads/master@{#9917}
