From af93fc08a19e75bc0b3fab393345c29b2094391d Mon Sep 17 00:00:00 2001 From: "kwiberg@webrtc.org" Date: Thu, 17 Jul 2014 08:18:33 +0000 Subject: [PATCH] AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter R=aluebs@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/audio_processing/audio_buffer.cc | 11 ----------- webrtc/modules/audio_processing/common.h | 2 +- 2 files changed, 1 insertion(+), 12 deletions(-) diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc index 35e1eb7c0b..7f579b0d92 100644 --- a/webrtc/modules/audio_processing/audio_buffer.cc +++ b/webrtc/modules/audio_processing/audio_buffer.cc @@ -285,7 +285,6 @@ void AudioBuffer::InitForNewData() { } const int16_t* AudioBuffer::data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return channels_->ibuf()->channel(channel); } @@ -295,7 +294,6 @@ int16_t* AudioBuffer::data(int channel) { } const float* AudioBuffer::data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return channels_->fbuf()->channel(channel); } @@ -305,7 +303,6 @@ float* AudioBuffer::data_f(int channel) { } const int16_t* AudioBuffer::low_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->low_channel(channel) : data(channel); } @@ -316,7 +313,6 @@ int16_t* AudioBuffer::low_pass_split_data(int channel) { } const float* AudioBuffer::low_pass_split_data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->low_channel_f(channel) : data_f(channel); } @@ -327,7 +323,6 @@ float* AudioBuffer::low_pass_split_data_f(int channel) { } const int16_t* AudioBuffer::high_pass_split_data(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->high_channel(channel) : NULL; } @@ -337,7 +332,6 @@ int16_t* AudioBuffer::high_pass_split_data(int channel) { } const float* AudioBuffer::high_pass_split_data_f(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); return split_channels_.get() ? split_channels_->high_channel_f(channel) : NULL; } @@ -348,19 +342,14 @@ float* AudioBuffer::high_pass_split_data_f(int channel) { } const int16_t* AudioBuffer::mixed_data(int channel) const { - assert(channel >= 0 && channel < num_mixed_channels_); - return mixed_channels_->channel(channel); } const int16_t* AudioBuffer::mixed_low_pass_data(int channel) const { - assert(channel >= 0 && channel < num_mixed_low_pass_channels_); - return mixed_low_pass_channels_->channel(channel); } const int16_t* AudioBuffer::low_pass_reference(int channel) const { - assert(channel >= 0 && channel < num_proc_channels_); if (!reference_copied_) { return NULL; } diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h index 42454df299..10249cc2bb 100644 --- a/webrtc/modules/audio_processing/common.h +++ b/webrtc/modules/audio_processing/common.h @@ -55,7 +55,7 @@ class ChannelBuffer { T* data() { return data_.get(); } T* channel(int i) { - assert(i < num_channels_); + assert(i >= 0 && i < num_channels_); return channels_[i]; } T** channels() { return channels_.get(); }