magjed 64e753c399 Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388

Sample output:
[ RUN      ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib:  extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
  Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[  FAILED  ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam =  and GetParam() =  (361 ms)

Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7a

TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1253573005

Cr-Commit-Position: refs/heads/master@{#9621}
2015-07-23 11:30:14 +00:00

183 lines
6.0 KiB
C++

/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/wav_file.h"
#include <algorithm>
#include <cstdio>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/wav_header.h"
namespace webrtc {
// We write 16-bit PCM WAV files.
static const WavFormat kWavFormat = kWavFormatPcm;
static const int kBytesPerSample = 2;
// Doesn't take ownership of the file handle and won't close it.
class ReadableWavFile : public ReadableWav {
public:
explicit ReadableWavFile(FILE* file) : file_(file) {}
virtual size_t Read(void* buf, size_t num_bytes) {
return fread(buf, 1, num_bytes, file_);
}
private:
FILE* file_;
};
WavReader::WavReader(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "rb")) {
CHECK(file_handle_ && "Could not open wav file for reading.");
ReadableWavFile readable(file_handle_);
WavFormat format;
int bytes_per_sample;
CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
&bytes_per_sample, &num_samples_));
num_samples_remaining_ = num_samples_;
CHECK_EQ(kWavFormat, format);
CHECK_EQ(kBytesPerSample, bytes_per_sample);
}
WavReader::~WavReader() {
Close();
}
size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to big-endian when reading from WAV file"
#endif
// There could be metadata after the audio; ensure we don't read it.
num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
num_samples_remaining_);
const size_t read =
fread(samples, sizeof(*samples), num_samples, file_handle_);
// If we didn't read what was requested, ensure we've reached the EOF.
CHECK(read == num_samples || feof(file_handle_));
CHECK_LE(read, num_samples_remaining_);
num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
return read;
}
size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
static const size_t kChunksize = 4096 / sizeof(uint16_t);
size_t read = 0;
for (size_t i = 0; i < num_samples; i += kChunksize) {
int16_t isamples[kChunksize];
size_t chunk = std::min(kChunksize, num_samples - i);
chunk = ReadSamples(chunk, isamples);
for (size_t j = 0; j < chunk; ++j)
samples[i + j] = isamples[j];
read += chunk;
}
return read;
}
void WavReader::Close() {
CHECK_EQ(0, fclose(file_handle_));
file_handle_ = NULL;
}
WavWriter::WavWriter(const std::string& filename, int sample_rate,
int num_channels)
: sample_rate_(sample_rate),
num_channels_(num_channels),
num_samples_(0),
file_handle_(fopen(filename.c_str(), "wb")) {
CHECK(file_handle_ && "Could not open wav file for writing.");
CHECK(CheckWavParameters(num_channels_,
sample_rate_,
kWavFormat,
kBytesPerSample,
num_samples_));
// Write a blank placeholder header, since we need to know the total number
// of samples before we can fill in the real data.
static const uint8_t blank_header[kWavHeaderSize] = {0};
CHECK_EQ(1u, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
}
WavWriter::~WavWriter() {
Close();
}
void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to WAV file"
#endif
const size_t written =
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
CHECK_EQ(num_samples, written);
num_samples_ += static_cast<uint32_t>(written);
CHECK(written <= std::numeric_limits<uint32_t>::max() ||
num_samples_ >= written); // detect uint32_t overflow
CHECK(CheckWavParameters(num_channels_,
sample_rate_,
kWavFormat,
kBytesPerSample,
num_samples_));
}
void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
static const size_t kChunksize = 4096 / sizeof(uint16_t);
for (size_t i = 0; i < num_samples; i += kChunksize) {
int16_t isamples[kChunksize];
const size_t chunk = std::min(kChunksize, num_samples - i);
FloatS16ToS16(samples + i, chunk, isamples);
WriteSamples(isamples, chunk);
}
}
void WavWriter::Close() {
CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
uint8_t header[kWavHeaderSize];
WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_);
CHECK_EQ(1u, fwrite(header, kWavHeaderSize, 1, file_handle_));
CHECK_EQ(0, fclose(file_handle_));
file_handle_ = NULL;
}
} // namespace webrtc
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
int num_channels) {
return reinterpret_cast<rtc_WavWriter*>(
new webrtc::WavWriter(filename, sample_rate, num_channels));
}
void rtc_WavClose(rtc_WavWriter* wf) {
delete reinterpret_cast<webrtc::WavWriter*>(wf);
}
void rtc_WavWriteSamples(rtc_WavWriter* wf,
const float* samples,
size_t num_samples) {
reinterpret_cast<webrtc::WavWriter*>(wf)->WriteSamples(samples, num_samples);
}
int rtc_WavSampleRate(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
}
int rtc_WavNumChannels(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
}
uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
}