1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "audio/remix_resample.h"
12
13 #include "api/audio/audio_frame.h"
14 #include "audio/utility/audio_frame_operations.h"
15 #include "common_audio/resampler/include/push_resampler.h"
16 #include "rtc_base/checks.h"
17
18 namespace webrtc {
19 namespace voe {
20
RemixAndResample(const AudioFrame & src_frame,PushResampler<int16_t> * resampler,AudioFrame * dst_frame)21 void RemixAndResample(const AudioFrame& src_frame,
22 PushResampler<int16_t>* resampler,
23 AudioFrame* dst_frame) {
24 RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
25 src_frame.num_channels_, src_frame.sample_rate_hz_,
26 resampler, dst_frame);
27 dst_frame->timestamp_ = src_frame.timestamp_;
28 dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
29 dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
30 dst_frame->packet_infos_ = src_frame.packet_infos_;
31 }
32
RemixAndResample(const int16_t * src_data,size_t samples_per_channel,size_t num_channels,int sample_rate_hz,PushResampler<int16_t> * resampler,AudioFrame * dst_frame)33 void RemixAndResample(const int16_t* src_data,
34 size_t samples_per_channel,
35 size_t num_channels,
36 int sample_rate_hz,
37 PushResampler<int16_t>* resampler,
38 AudioFrame* dst_frame) {
39 const int16_t* audio_ptr = src_data;
40 size_t audio_ptr_num_channels = num_channels;
41 int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
42
43 // Downmix before resampling.
44 if (num_channels > dst_frame->num_channels_) {
45 RTC_DCHECK(num_channels == 2 || num_channels == 4)
46 << "num_channels: " << num_channels;
47 RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
48 << "dst_frame->num_channels_: " << dst_frame->num_channels_;
49
50 AudioFrameOperations::DownmixChannels(
51 src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
52 downmixed_audio);
53 audio_ptr = downmixed_audio;
54 audio_ptr_num_channels = dst_frame->num_channels_;
55 }
56
57 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
58 audio_ptr_num_channels) == -1) {
59 RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
60 << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
61 << dst_frame->sample_rate_hz_
62 << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
63 }
64
65 // TODO(yujo): for muted input frames, don't resample. Either 1) allow
66 // resampler to return output length without doing the resample, so we know
67 // how much to zero here; or 2) make resampler accept a hint that the input is
68 // zeroed.
69 const size_t src_length = samples_per_channel * audio_ptr_num_channels;
70 int out_length =
71 resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
72 AudioFrame::kMaxDataSizeSamples);
73 if (out_length == -1) {
74 RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
75 << ", src_length = " << src_length
76 << ", dst_frame->mutable_data() = "
77 << dst_frame->mutable_data();
78 }
79 dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
80
81 // Upmix after resampling.
82 if (num_channels == 1 && dst_frame->num_channels_ == 2) {
83 // The audio in dst_frame really is mono at this point; MonoToStereo will
84 // set this back to stereo.
85 dst_frame->num_channels_ = 1;
86 AudioFrameOperations::UpmixChannels(2, dst_frame);
87 }
88 }
89
90 } // namespace voe
91 } // namespace webrtc
92