This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer
flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes
compile error for MIPS, as some MIPS optimization blocks use maximum possible
number of available registers.
Also, this change contains minor GN build fix for MIPS platform regarding the
pitch_filter_mips.c / pitch_filter_c.c file inclusion.
BUG=477749
R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48139004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
Cr-Commit-Position: refs/heads/master@{#9047}
Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.
BUG=2692
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51559004
Cr-Commit-Position: refs/heads/master@{#9025}
With this change, the currently used encoder is held in a scoped_ptr.
iSAC is a special case, since the encoder instance is also a decoder
instance, so it may have to be available also if another send codec is
used. This is accomplished by having a separate scoped_ptr for iSAC.
Remove mirror ID from ACM codec database functions, and remove unused
functions from the database.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48729004
Cr-Commit-Position: refs/heads/master@{#8982}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
- removed commented code lines used during development
- excluded fft.c since there are neon optimizations used and a removal may cause a performance regression
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48799004
Cr-Commit-Position: refs/heads/master@{#8967}
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.
BUG=chromium:440793
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119005
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8963}
This reverts commit cf3c83e76c273309558c86fda915410f65b7a899.
Reverting EventWrapper split did not fix the issue, re-landing.
BUG=chromium:470013
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49629004
Cr-Commit-Position: refs/heads/master@{#8946}
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
hence trivial.
The macro name may in fact mislead the user to assume a cast/truncation to int32_t is done.
- Removing usage of it.
- Some style changes.
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46749005
Cr-Commit-Position: refs/heads/master@{#8918}
This reverts commit 9509fbfc301dd5412804ce5731afedc81480f2f8.
This is to debug a Chromium issue that WebRTC hangs if there is > 1 PeerConnection active in the browser on Win XP.
BUG=
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43019004
Cr-Commit-Position: refs/heads/master@{#8912}
Pass content_browsertests in Chromium. Performance test result (lower is
better):
C version: 100%
old intrinsics Neon version (with bug): 16.5%
new intrinsics Neon version: 18.0%
asm Neon version: 23.3%
BUG=4002
R=andrew@webrtc.org, jridges@masque.com
Change-Id: Ia0a96ac237216b635fc528f67d39319cdf246281
Review URL: https://webrtc-codereview.appspot.com/46739004
Cr-Commit-Position: refs/heads/master@{#8907}
This change essentially divides AudioCodingModuleImpl into two parts:
one is the code related to managing codecs, now moved into CodecManager,
and the other is what remains in AudioCodingModuleImpl.
This change also removes AudioCodingModuleImpl::InitializeSender. The
function was essentially no-op, since it was always called immediately
after construction.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51469004
Cr-Commit-Position: refs/heads/master@{#8893}
We want to crate the illusion that iSAC supports 48000 Hz decoding,
while in fact it outputs 32000 Hz. This is the iSAC fullband mode.
Currently this is (also) handled by higher layers, but in future
changes this will not be the case.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47809004
Cr-Commit-Position: refs/heads/master@{#8889}
Change internal indexing of registered decoders. It makes sense because payload type is unique, while ACM codec ID may not be. This is a step towards allowing for addition of external decoders.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44869004
Cr-Commit-Position: refs/heads/master@{#8867}
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.
This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.
R=mflodman@webrtc.org, mflodman
BUG=
Review URL: https://webrtc-codereview.appspot.com/48599004
Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
In SDP, RED audio codec has its own sample rate. Currently, we offer RED/8000 (8 kHz). But the actual send codec can violate this sample rate. The way to solve it is to introduce more RED payload types, e.g., RED/16000, RED/32000.
As a first step towards that, we, in this CL, limit the current RED (RED/8000) to work only with 8 kHz codecs.
BUG=3619
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43849004
Cr-Commit-Position: refs/heads/master@{#8830}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8830 4adac7df-926f-26a2-2b94-8c16560cd09d
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).
BUG=4423
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42849004
Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
Also removes unnecessary casts to int32_t from int16_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48519004
Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
Move timestamp conversion out of ACMGenericCodec. Also remove lock
from ACMGenericCodec since the instance is always protected by
acm_crit_sect_ in AudioCodingModuleImpl.
Restructuring the code in AudioCodingModuleImpl::Encode to streamline
the use of locks.
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46479004
Cr-Commit-Position: refs/heads/master@{#8773}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8773 4adac7df-926f-26a2-2b94-8c16560cd09d
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.
The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.
BUG=4361
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45619004
Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see
common_audio/signal_processing/include/spl_inl_armv7.h and
common_audio/signal_processing/include/spl_inl_mips.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int
or int32_t)
Some other minor code cleanup also exists.
BUG=3348,3353
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42639004
Cr-Commit-Position: refs/heads/master@{#8717}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8717 4adac7df-926f-26a2-2b94-8c16560cd09d
The current way that iSAC RCU is packetized and sent as a RED packet,
with the same payload type for primary and redundant payloads, does
not follow the specification for RED. As it is now, it is impossible
for a receiver to know if an incoming RED packet with iSAC payloads
inside consists of two "primary" (but time-shifted) payloads, or one
primary and one RCU payload. The RED standard stipulates that the
former option is the correct interpretation, while our implementation
currently applies the latter.
This CL removes support for iSAC RCU from Audio Coding Module, but
leaves it in the iSAC codec itself (i.e., in the C implementation).
BUG=4402
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45569004
Cr-Commit-Position: refs/heads/master@{#8713}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8713 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call.
Unit tests were updated to use the new method.
Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation.
Other refactoring work that was done, that may not be obvious why:
1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive().
2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40259005
Cr-Commit-Position: refs/heads/master@{#8671}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
In addition an implicit cast from int32_t to int16_t was removed, which was a bug.
BUG=3348, 3353
TESTED=Locally on Mac and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41179004
Cr-Commit-Position: refs/heads/master@{#8653}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8653 4adac7df-926f-26a2-2b94-8c16560cd09d
