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Consolidate audio conversion from Channel and TransmitMixer.

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Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.

Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.

BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.

R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-03 21:56:01 +00:00
parent cca888a5bf
commit 40ee3d07ed
18 changed files with 1663 additions and 1791 deletions
Download Patch File Download Diff File Expand all files Collapse all files

42
webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
View File

@@ -13,20 +13,15 @@
#include <string.h>
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/logging.h"
namespace webrtc {
namespace acm2 {
ACMResampler::ACMResampler()
: resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
ACMResampler::ACMResampler() {
}
ACMResampler::~ACMResampler() {
delete resampler_crit_sect_;
}
int ACMResampler::Resample10Msec(const int16_t* in_audio,
@@ -34,37 +29,28 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
int out_freq_hz,
int num_audio_channels,
int16_t* out_audio) {
CriticalSectionScoped cs(resampler_crit_sect_);
int in_length = in_freq_hz * num_audio_channels / 100;
int out_length = out_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
memcpy(out_audio, in_audio, length * sizeof(int16_t));
return static_cast<int16_t>(in_freq_hz / 100);
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
return in_length / num_audio_channels;
}
// |maxLen| is maximum number of samples for 10ms at 48kHz.
int max_len = 480 * num_audio_channels;
int length_in = (in_freq_hz / 100) * num_audio_channels;
int out_len;
ResamplerType type = (num_audio_channels == 1) ? kResamplerSynchronous :
kResamplerSynchronousStereo;
if (resampler_.ResetIfNeeded(in_freq_hz, out_freq_hz, type) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
"Error in reset of resampler");
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
num_audio_channels) != 0) {
LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
num_audio_channels);
return -1;
}
if (resampler_.Push(in_audio, length_in, out_audio, max_len, out_len) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
"Error in resampler: resampler.Push");
out_length = resampler_.Resample(in_audio, in_length, out_audio, out_length);
if (out_length == -1) {
LOG_FERR4(LS_ERROR, Resample, in_audio, in_length, out_audio, out_length);
return -1;
}
return out_len / num_audio_channels;
return out_length / num_audio_channels;
}
} // namespace acm2
} // namespace webrtc

10
webrtc/modules/audio_coding/main/acm2/acm_resampler.h
View File

@@ -11,13 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
namespace acm2 {
class ACMResampler {
@@ -32,13 +29,10 @@ class ACMResampler {
int16_t* out_audio);
private:
// Use the Resampler class.
Resampler resampler_;
CriticalSectionWrapper* resampler_crit_sect_;
PushResampler resampler_;
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_

8
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@@ -1205,11 +1205,7 @@ int AudioCodingModuleImpl::Add10MsData(
return -1;
}
// Allow for 8, 16, 32 and 48kHz input audio.
if ((audio_frame.sample_rate_hz_ != 8000)
&& (audio_frame.sample_rate_hz_ != 16000)
&& (audio_frame.sample_rate_hz_ != 32000)
&& (audio_frame.sample_rate_hz_ != 48000)) {
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot Add 10 ms audio, input frequency not valid");
@@ -1371,7 +1367,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
if (preprocess_frame_.samples_per_channel_ < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot add 10 ms audio, resmapling failed");
"Cannot add 10 ms audio, resampling failed");
return -1;
}
preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq;

8
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@@ -1273,11 +1273,7 @@ int32_t AudioCodingModuleImpl::Add10MsData(
return -1;
}
// Allow for 8, 16, 32 and 48kHz input audio.
if ((audio_frame.sample_rate_hz_ != 8000)
&& (audio_frame.sample_rate_hz_ != 16000)
&& (audio_frame.sample_rate_hz_ != 32000)
&& (audio_frame.sample_rate_hz_ != 48000)) {
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot Add 10 ms audio, input frequency not valid");
@@ -1444,7 +1440,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
if (preprocess_frame_.samples_per_channel_ < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot add 10 ms audio, resmapling failed");
"Cannot add 10 ms audio, resampling failed");
return -1;
}
preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq;