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Bug 1404997 - P23. Strongly enforced that our destination buffer is big enough. r=padenot

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MozReview-Commit-ID: A1kLsH75SzX

--HG--
extra : rebase_source : bcc4460507638ef36986ad519e48ba2d4596f4cf
This commit is contained in:
Jean-Yves Avenard 2017-12-13 18:44:30 +01:00
parent 485b10a8b0
commit 1769747959
3 changed files with 10 additions and 9 deletions
repo.diff.show_diff_stats Download patch file Download diff file Expand all files Collapse all files

9
media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
View file

@ -715,7 +715,6 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
{
CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
unsigned int numSamples = 0;
//validate params
if(!speechData )
@ -726,7 +725,7 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
}
// Validate sample length
if((numSamples = GetNum10msSamplesForFrequency(samplingFreqHz)) == 0 )
if(GetNum10msSamplesForFrequency(samplingFreqHz) == 0)
{
CSFLogError(LOGTAG,"%s Invalid Sampling Frequency ", __FUNCTION__);
MOZ_ASSERT(PR_FALSE);
@ -749,7 +748,7 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
return kMediaConduitSessionNotInited;
}
int lengthSamplesAllowed = lengthSamples;
lengthSamples = 0; //output paramter
if (mPtrVoEXmedia->GetAudioFrame(mChannel,
@ -766,8 +765,8 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
// XXX Annoying, have to copy to our buffers -- refactor?
lengthSamples = mAudioFrame.samples_per_channel_ * mAudioFrame.num_channels_;
PodCopy(speechData, mAudioFrame.data_,
lengthSamples);
MOZ_RELEASE_ASSERT(lengthSamples <= lengthSamplesAllowed);
PodCopy(speechData, mAudioFrame.data_, lengthSamples);
// Not #ifdef DEBUG or on a log module so we can use it for about:webrtc/etc
mSamples += lengthSamples;

1
media/webrtc/signaling/src/media-conduit/AudioConduit.h
View file

@ -140,6 +140,7 @@ public:
* @param speechData [in]: Pointer to a array to which a 10ms frame of audio will be copied
* @param samplingFreqHz [in]: Frequency of the sampling for playback in Hertz (16000, 32000,..)
* @param capture_delay [in]: Estimated Time between reading of the samples to rendering/playback
* @param lengthSamples [in]: Contain maximum length of speechData array.
* @param lengthSamples [out]: Will contain length of the audio frame in samples at return.
Ex: A value of 160 implies 160 samples each of 16-bits was copied
into speechData

9
media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
View file

@ -2221,9 +2221,11 @@ private:
TrackTicks framesNeeded = aDesiredTime - mPlayedTicks;
while (framesNeeded >= 0) {
int16_t scratchBuffer[AUDIO_SAMPLE_BUFFER_MAX_BYTES / sizeof(int16_t)];
const int scratchBufferLength =
AUDIO_SAMPLE_BUFFER_MAX_BYTES / sizeof(int16_t);
int16_t scratchBuffer[scratchBufferLength];
int samplesLength;
int samplesLength = scratchBufferLength;
// This fetches 10ms of data, either mono or stereo
MediaConduitErrorCode err =
@ -2248,8 +2250,7 @@ private:
PodArrayZero(scratchBuffer);
}
MOZ_ASSERT(samplesLength * sizeof(uint16_t) <=
AUDIO_SAMPLE_BUFFER_MAX_BYTES);
MOZ_RELEASE_ASSERT(samplesLength <= scratchBufferLength);
CSFLogDebug(
LOGTAG, "Audio conduit returned buffer of length %u", samplesLength);